1 === release 1.15.90 ===
3 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
9 * gst-rtsp-server.doap:
13 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
15 * gst/rtsp-server/rtsp-stream.c:
16 rtsp-stream: Add support for GCM (RFC 7714)
19 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
21 * gst/rtsp-server/rtsp-session-pool.c:
22 session pool: fix missing klass-> in klass->create_session
24 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
27 g-i: pass --quiet to g-ir-scanner
28 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
29 that we get even if everything works just fine.
30 We still get g-ir-scanner warnings and compiler warnings if
33 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
36 g-i: silence 'nested extern' compiler warnings when building scanner binary
37 We need a nested extern in our init section for the scanner binary
38 so we can call gst_init to make sure GStreamer types are initialised
39 (they are not all lazy init via get_type functions, but some are in
40 exported variables). There doesn't seem to be any other mechanism to
41 achieve this, so just remove that warning, it's not important at all.
43 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
46 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
48 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
50 * gst/rtsp-server/rtsp-media.c:
51 * tests/check/gst/media.c:
52 rtsp-media: Handle set state when preparing.
53 Handle the situation when a call to gst_rtsp_media_set_state is done
54 when media status is preparing.
55 Also add unit test for this scenario.
56 The unit test simulate on a media level when two clients share a (live)
58 Both clients have done SETUP and got responses. Now client 1 is doing
59 play and client 2 is just closing the connection.
60 Then without patch there are a problem when
61 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
62 And client2 is doing closing connection we can end up in a call
63 to gst_rtsp_media_set_state when
64 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
65 shut down media is jumped over .
66 With this patch and this scenario we wait until
67 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
68 execute after that and now we will execute the logic for
71 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
79 === release 1.15.2 ===
81 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
87 * gst-rtsp-server.doap:
91 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
93 * gst/rtsp-server/rtsp-media.c:
94 * tests/check/gst/client.c:
95 rtsp-media: Fix multicast use case with common media
104 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
106 * gst/rtsp-server/rtsp-client.c:
107 * gst/rtsp-server/rtsp-stream.c:
108 * gst/rtsp-server/rtsp-stream.h:
109 rtsp-server: remove recursive behavior
110 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
112 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
114 * gst/rtsp-server/rtsp-client.c:
115 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
116 And route all messages through the send_func if no send_messages_func
118 We otherwise break backwards compatibility.
120 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
122 * docs/libs/gst-rtsp-server-sections.txt:
123 * gst/rtsp-server/rtsp-client.c:
124 * gst/rtsp-server/rtsp-client.h:
125 * gst/rtsp-server/rtsp-stream.c:
126 rtsp-client: Add support for sending buffer lists directly
127 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
129 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
131 * docs/libs/gst-rtsp-server-sections.txt:
132 * gst/rtsp-server/rtsp-client.c:
133 * gst/rtsp-server/rtsp-media.c:
134 * gst/rtsp-server/rtsp-stream-transport.c:
135 * gst/rtsp-server/rtsp-stream-transport.h:
136 * gst/rtsp-server/rtsp-stream.c:
137 * gst/rtsp-sink/gstrtspclientsink.c:
138 rtsp-server: Add support for buffer lists
139 This adds new functions for passing buffer lists through the different
140 layers without breaking API/ABI, and enables the appsink to actually
141 provide buffer lists.
142 This should already reduce CPU usage and potentially context switches a
143 bit by passing a whole buffer list from the appsink instead of
144 individual buffers. As a next step it would be necessary to
145 a) Add support for a vector of data for the GstRTSPMessage body
146 b) Add support for sending multiple messages at once to the
147 GstRTSPWatch and let it be handled internally
148 c) Adding API to GOutputStream that works like writev()
149 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
151 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
153 * gst/rtsp-server/rtsp-client.c:
154 client: Fix crash in close handler
155 The close handler could trigger a crash because it invalidated the
156 watch_context while still leaving a source attached to it which would be
157 cleaned up at a later point.
159 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
161 * gst/rtsp-server/rtsp-stream.c:
162 rtsp-stream: Use cached address when allocating sockets
163 If an address/port was previously decided upon (ex: multicast in the
164 SDP), then use that instead of re-creating another one
165 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
167 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
169 * gst/rtsp-server/rtsp-media.c:
170 rtsp-media: Fix race codition in finish_unprepare
171 The previous fix for race condition around finish_unprepare where the
172 function could be called twice assumed that the status wouldn't change
173 during execution of the function. This assumption is incorrect as the
174 state may change, for example if an error message arrives from the
176 Instead a flag keeping track on whether the finish_unprepare function
177 is currently executing is introduced and checked.
178 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
180 === release 1.15.1 ===
182 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
188 * gst-rtsp-server.doap:
192 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
194 * gst/rtsp-server/rtsp-stream.c:
195 Add source elements to the pipeline before activation
196 In plug_src we changed the element state before adding it to
197 the owner container. This prevented the pipeline from intercepting
198 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
199 to assign a custom task pool.
200 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
202 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
205 Automatic update of common submodule
206 From ed78bee to 59cb678
208 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
210 * examples/test-appsrc.c:
211 examples: test-appsrc: fix coding style error
213 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
215 * examples/test-appsrc.c:
216 examples: test-appsrc: fix buffer leak
218 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
220 * gst/rtsp-server/rtsp-media.c:
221 rtsp-media: Update priv->blocked when linked streams are unblocked.
222 Media is considered to be blocked when all streams that belong to
223 that media are blocked.
224 This patch solves the problem of inconsistent updates of
225 priv->blocked that are not synchronized with the media state.
227 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
229 * gst/rtsp-server/rtsp-media.c:
230 rtsp-media: Don't block streams before seeking
231 Before the seek operation is performed on media, it's required that
232 its pipeline is prepared <=> the pipeline is in the PAUSED state.
233 At this stage, all transport parts (transport sinks) have been successfully
234 added to the pipeline and there is no need for blocking the streams.
236 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
238 * tests/check/gst/rtspserver.c:
239 tests: rtspserver: Add shared media test case for TCP
241 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
243 * gst/rtsp-server/rtsp-stream.c:
244 rtsp-stream: Use seqnum-offset for rtpinfo
245 The sequence number in the rtpinfo is supposed to be the first RTP
246 sequence number. The "seqnum" property on a payloader is supposed to be
247 the number from the last processed RTP packet. The sequence number for
248 payloaders that inherit gstrtpbasepayload will not be correct in case of
249 buffer lists. In order to fix the seqnum property on the payloaders
250 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
251 "seqnum-offset" from the "stats" property contains the value of the
252 very first RTP packet in a stream. The server will, however, try to look
253 at the last simple in the sink element and only use properties on the
254 payloader in case there no sink elements yet, and by looking at the last
255 sample of the sink gives the server full control of which RTP packet it
256 looks at. If the payloader does not have the "stats" property, "seqnum"
257 is still used since "seqnum-offset" is only present in as part of
258 "stats" and this is still an issue not solved with this patch.
259 Needed for gst-plugins-base!17
261 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
263 * gst/rtsp-server/rtsp-stream.c:
264 rtsp-stream: Plug memory leak
265 Attaching a GSource to a context will increase the refcount. The idle
266 source will never be free'd since the initial reference is never
269 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
272 Add Gitlab CI configuration
273 This commit adds a .gitlab-ci.yml file, which uses a feature
274 to fetch the config from a centralized repository. The intent is
275 to have all the gstreamer modules use the same configuration.
276 The configuration is currently hosted at the gst-ci repository
277 under the gitlab/ci_template.yml path.
278 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
280 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
283 * gst-rtsp-server.doap:
284 Update git locations to gitlab
286 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
288 * gst/rtsp-server/meson.build:
289 meson: add new onvif types
291 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
293 * gst/rtsp-server/meson.build:
294 Add ONVIF subclass headers to the installed headers in meson.build too
296 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
298 * gst/rtsp-server/rtsp-server-object.h:
299 * gst/rtsp-server/rtsp-server.h:
300 rtsp-server: Declare GstRTSPServer struct before anything else
301 It's needed by all kinds of other headers, including the ones that are
302 required for defining the GstRTSPServer struct itself and its API.
304 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
306 * gst/rtsp-server/rtsp-onvif-client.h:
307 * gst/rtsp-server/rtsp-onvif-media-factory.h:
308 * gst/rtsp-server/rtsp-onvif-media.h:
309 * gst/rtsp-server/rtsp-onvif-server.h:
310 Mark all ONVIF-specific subclasses as Since 1.14
312 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
314 * gst/rtsp-server/Makefile.am:
315 * gst/rtsp-server/meson.build:
316 * gst/rtsp-server/rtsp-context.h:
317 * gst/rtsp-server/rtsp-onvif-server.c:
318 * gst/rtsp-server/rtsp-onvif-server.h:
319 * gst/rtsp-server/rtsp-server-object.h:
320 * gst/rtsp-server/rtsp-server-prelude.h:
321 * gst/rtsp-server/rtsp-server.c:
322 * gst/rtsp-server/rtsp-server.h:
323 * gst/rtsp-server/rtsp-session.h:
324 Include ONVIF types from single-include rtsp-server.h
325 ... by actually making it a single-include header and moving everything
326 related to the GstRTSPServer type to rtsp-server-object.h instead.
327 Otherwise there are too many circular includes.
328 https://bugzilla.gnome.org/show_bug.cgi?id=797361
330 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
332 * gst/rtsp-server/rtsp-client.c:
333 * gst/rtsp-server/rtsp-latency-bin.c:
334 * gst/rtsp-server/rtsp-stream.c:
335 * gst/rtsp-server/rtsp-stream.h:
336 rtsp-stream: use idle source in on_message_sent
337 When the underlying layers are running on_message_sent, this sometimes
338 causes the underlying layer to send more data, which will cause the
339 underlying layer to run callback on_message_sent again. This can go on
341 To break this chain, we introduce an idle source that takes care of
342 sending data if there are more to send when running callback
343 https://bugzilla.gnome.org/show_bug.cgi?id=797289
345 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
347 * gst/rtsp-server/rtsp-client.c:
348 rtsp-client: Remove timeout GSource on cleanup
349 Avoids ending up with races where a timeout would still be around
350 *after* a client was gone. This could happen rather easily in
351 RTSP-over-HTTP mode on a local connection, where each RTSP message
352 would be sent as a different HTTP connection with the same tunnelid.
353 If not properly removed, that timeout would then try to free again
354 a client (and its contents).
356 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
358 * gst/rtsp-server/Makefile.am:
359 autotools: fix distcheck
361 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
363 * gst/rtsp-server/Makefile.am:
364 * gst/rtsp-server/meson.build:
365 * gst/rtsp-server/rtsp-latency-bin.c:
366 * gst/rtsp-server/rtsp-latency-bin.h:
367 * gst/rtsp-server/rtsp-onvif-media.c:
368 onvif: encapsulate onvif part into a bin
369 ...and thus do not let onvif affect pipelines latency
370 https://bugzilla.gnome.org/show_bug.cgi?id=797174
372 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
374 * tests/check/gst/client.c:
375 tests: client: Avoid bind() failures in tests
376 https://bugzilla.gnome.org/show_bug.cgi?id=797059
378 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
380 * gst/rtsp-server/rtsp-media-factory.c:
381 * gst/rtsp-server/rtsp-media-factory.h:
382 * gst/rtsp-server/rtsp-media.c:
383 * gst/rtsp-server/rtsp-media.h:
384 * gst/rtsp-server/rtsp-stream.c:
385 * gst/rtsp-server/rtsp-stream.h:
386 * tests/check/gst/client.c:
387 * tests/check/gst/mediafactory.c:
388 New property for socket binding to mcast addresses
389 By default the multicast sockets are bound to INADDR_ANY,
390 as it's not allowed to bind sockets to multicast addresses
391 in Windows. This default behaviour can be changed by setting
392 bind-mcast-address property on the media-factory object.
393 https://bugzilla.gnome.org/show_bug.cgi?id=797059
395 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
398 * gst/rtsp-server/Makefile.am:
399 * gst/rtsp-server/meson.build:
400 * gst/rtsp-server/rtsp-address-pool.c:
401 * gst/rtsp-server/rtsp-auth.c:
402 * gst/rtsp-server/rtsp-client.c:
403 * gst/rtsp-server/rtsp-context.c:
404 * gst/rtsp-server/rtsp-media-factory-uri.c:
405 * gst/rtsp-server/rtsp-media-factory.c:
406 * gst/rtsp-server/rtsp-media.c:
407 * gst/rtsp-server/rtsp-mount-points.c:
408 * gst/rtsp-server/rtsp-params.c:
409 * gst/rtsp-server/rtsp-permissions.c:
410 * gst/rtsp-server/rtsp-sdp.c:
411 * gst/rtsp-server/rtsp-server-prelude.h:
412 * gst/rtsp-server/rtsp-server.c:
413 * gst/rtsp-server/rtsp-session-media.c:
414 * gst/rtsp-server/rtsp-session-pool.c:
415 * gst/rtsp-server/rtsp-session.c:
416 * gst/rtsp-server/rtsp-stream-transport.c:
417 * gst/rtsp-server/rtsp-stream.c:
418 * gst/rtsp-server/rtsp-thread-pool.c:
419 * gst/rtsp-server/rtsp-token.c:
421 libs: fix API export/import and 'inconsistent linkage' on MSVC
422 Export rtsp-server library API in headers when we're building the
423 library itself, otherwise import the API from the headers.
424 This fixes linker warnings on Windows when building with MSVC.
425 Fix up some missing config.h includes when building the lib which
426 is needed to get the export api define from config.h
427 https://bugzilla.gnome.org/show_bug.cgi?id=797185
429 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
431 * gst/rtsp-server/rtsp-media-factory.c:
432 rtsp-media-factory: Add missing break statements
433 This resulted in warnings/assertions whenever one accessed the
434 max-mcast-ttl property.
438 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
442 meson: add gobject-cast-checks, glib-asserts, glib-checks options
444 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
448 * tests/check/meson.build:
449 meson: add option to disable build of rtspclientsink plugin
451 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
454 meson: re-arrange options
456 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
460 * tests/check/meson.build:
462 meson: Use feature option for tests option
463 This was somehow missed the last time around.
465 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
467 * gst/rtsp-server/meson.build:
469 meson: Maintain macOS ABI through dylib versioning
470 Requires Meson 0.48, but the feature will be ignored on older versions
471 so it's safe to add it without bumping the requirement.
473 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
475 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
477 * gst/rtsp-sink/meson.build:
479 meson: add pkg-config file for the rtspclientsink plugin
481 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
483 * gst/rtsp-server/rtsp-client.c:
484 * tests/check/gst/client.c:
485 rtsp-client: Avoid reuse of channel numbers for interleaved
486 If a (strange) client would reuse interleaved channel numbers in
487 multiple SETUP requests, we should not accept them. The channel
488 numbers are used for looking up stream transports in the
489 priv->transports hash table, and transports disappear from the table
490 if channel numbers are reused.
491 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
492 server to change the channel numbers suggested by the client.
493 https://bugzilla.gnome.org/show_bug.cgi?id=796988
495 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
497 * tests/check/gst/client.c:
498 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
499 Allow regex for matching transport header against expected pattern.
500 https://bugzilla.gnome.org/show_bug.cgi?id=796988
502 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
504 * tests/check/meson.build:
505 meson: There is no gstreamer-plugins-good-1.0.pc
506 There is no installed version of that, only an uninstalled version.
508 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
510 * gst/rtsp-server/rtsp-client.c:
511 * tests/check/gst/stream.c:
512 Fix indentation again
514 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
516 * gst/rtsp-server/rtsp-client.c:
517 * gst/rtsp-server/rtsp-stream.c:
518 * gst/rtsp-server/rtsp-stream.h:
519 * tests/check/gst/client.c:
520 * tests/check/gst/stream.c:
521 stream: Added a list of multicast client addresses
522 When media is shared, the same media stream can be sent
523 to multiple multicast groups. Currently, there is no API
524 to retrieve multicast addresses from the stream.
525 When calling gst_rtsp_stream_get_multicast_address() function,
526 only the first multicast address is returned.
527 With this patch, each multicast destination requested in SETUP
528 will be stored in an internal list (call to
529 gst_rtsp_stream_add_multicast_client_address()).
530 The list of multicast groups requested by the clients can be
531 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
532 There still exist some problems with the current implementation
533 in the multicast case:
534 1) The receiving part is currently only configured with
535 regard to the first multicast client (see
536 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
537 2) Secondly, of security reasons, some constraints should be
538 put on the requested multicast destinations (see
539 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
540 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
541 https://bugzilla.gnome.org/show_bug.cgi?id=793441
543 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
545 * gst/rtsp-server/rtsp-client.c:
546 * gst/rtsp-server/rtsp-stream.c:
547 * gst/rtsp-server/rtsp-stream.h:
548 * tests/check/gst/client.c:
549 stream: Choose the maximum ttl value provided by multicast clients
550 The maximum ttl value provided so far by the multicast clients
551 will be chosen and reported in the response to the current
553 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
554 https://bugzilla.gnome.org/show_bug.cgi?id=793441
556 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
558 * gst/rtsp-server/rtsp-stream.c:
559 * tests/check/gst/client.c:
560 rtsp-stream: Don't require address pool in the transport specific case
561 If "transport.client-settings" parameter is set to true, the client is
562 allowed to specify destination, ports and ttl.
563 There is no need for pre-configured address pool.
564 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
565 https://bugzilla.gnome.org/show_bug.cgi?id=793441
567 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
569 * gst/rtsp-server/rtsp-client.c:
570 * tests/check/gst/client.c:
571 client: Don't reserve multicast address in the client setting case
572 When two multicast clients request specific transport
573 configurations, and "transport.client-settings" parameter is
574 set to true, it's wrong to actually require that these two
575 clients request the same multicast group.
576 Removed test_client_multicast_invalid_transport_specific test
577 cases as they wrongly require that the requested destination
578 address is supposed to be present in the address pool, also in
579 the case when "transport.client-settings" parameter is set to true.
580 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
581 https://bugzilla.gnome.org/show_bug.cgi?id=793441
583 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
585 * gst/rtsp-server/rtsp-media-factory.c:
586 * gst/rtsp-server/rtsp-media-factory.h:
587 * gst/rtsp-server/rtsp-media.c:
588 * gst/rtsp-server/rtsp-media.h:
589 * gst/rtsp-server/rtsp-stream.c:
590 * gst/rtsp-server/rtsp-stream.h:
591 * tests/check/gst/mediafactory.c:
592 Add new API for setting/getting maximum multicast ttl value
593 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
594 https://bugzilla.gnome.org/show_bug.cgi?id=793441
596 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
598 * gst/rtsp-server/rtsp-stream.c:
599 rtsp-stream: avoid duplicating the first multicast client
600 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
601 clients were dynamically added and removed to the multicast
602 udp sinks, as such we should no longer add a first client in
603 set_multicast_socket_for_udpsink
604 https://bugzilla.gnome.org/show_bug.cgi?id=793441
606 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
608 * gst/rtsp-server/rtsp-stream.c:
609 Revert "rtsp-stream: avoid duplicating the first multicast client"
610 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
611 Commits where accidentially squashed together
613 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
615 * gst/rtsp-server/rtsp-client.c:
616 * gst/rtsp-server/rtsp-media-factory.c:
617 * gst/rtsp-server/rtsp-media-factory.h:
618 * gst/rtsp-server/rtsp-media.c:
619 * gst/rtsp-server/rtsp-media.h:
620 * gst/rtsp-server/rtsp-stream.c:
621 * gst/rtsp-server/rtsp-stream.h:
622 * tests/check/gst/client.c:
623 * tests/check/gst/mediafactory.c:
624 Revert "Add new API for setting/getting maximum multicast ttl value"
625 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
626 Commits where accidentially squashed together
628 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
630 * gst/rtsp-server/rtsp-stream.c:
631 * tests/check/gst/client.c:
632 Revert "rtsp-stream: Don't require address pool in the transport specific case"
633 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
634 Commits where accidentially squashed together
636 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
638 * gst/rtsp-server/rtsp-client.c:
639 * gst/rtsp-server/rtsp-stream.c:
640 * gst/rtsp-server/rtsp-stream.h:
641 * tests/check/gst/client.c:
642 * tests/check/gst/stream.c:
643 Revert "stream: Choose the maximum ttl value provided by multicast clients"
644 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
645 Commits where accidentially squashed together
647 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
649 * examples/test-auth-digest.c:
650 examples: Fix indentation
652 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
654 * gst/rtsp-server/rtsp-client.c:
655 * gst/rtsp-server/rtsp-stream.c:
656 * gst/rtsp-server/rtsp-stream.h:
657 * tests/check/gst/client.c:
658 * tests/check/gst/stream.c:
659 stream: Choose the maximum ttl value provided by multicast clients
660 The maximum ttl value provided so far by the multicast clients
661 will be chosen and reported in the response to the current
663 https://bugzilla.gnome.org/show_bug.cgi?id=793441
665 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
667 * gst/rtsp-server/rtsp-stream.c:
668 * tests/check/gst/client.c:
669 rtsp-stream: Don't require address pool in the transport specific case
670 If "transport.client-settings" parameter is set to true, the client is
671 allowed to specify destination, ports and ttl.
672 There is no need for pre-configured address pool.
673 https://bugzilla.gnome.org/show_bug.cgi?id=793441
675 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
677 * gst/rtsp-server/rtsp-client.c:
678 * gst/rtsp-server/rtsp-media-factory.c:
679 * gst/rtsp-server/rtsp-media-factory.h:
680 * gst/rtsp-server/rtsp-media.c:
681 * gst/rtsp-server/rtsp-media.h:
682 * gst/rtsp-server/rtsp-stream.c:
683 * gst/rtsp-server/rtsp-stream.h:
684 * tests/check/gst/client.c:
685 * tests/check/gst/mediafactory.c:
686 Add new API for setting/getting maximum multicast ttl value
687 https://bugzilla.gnome.org/show_bug.cgi?id=793441
689 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
691 * gst/rtsp-server/rtsp-stream.c:
692 rtsp-stream: avoid duplicating the first multicast client
693 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
694 clients were dynamically added and removed to the multicast
695 udp sinks, as such we should no longer add a first client in
696 set_multicast_socket_for_udpsink
697 https://bugzilla.gnome.org/show_bug.cgi?id=793441
699 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
701 * gst/rtsp-server/Makefile.am:
702 rtsp-server: Add gstreamer-base gir dir in autotools
704 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
706 * gst/rtsp-server/rtsp-client.c:
707 * gst/rtsp-server/rtsp-stream.c:
708 rtsp-client: always allocate both IPV4 and IPV6 sockets
709 multiudpsink does not support setting the socket* properties
710 after it has started, which meant that rtsp-server could no
711 longer serve on both IPV4 and IPV6 sockets since the patches
712 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
714 When first connecting an IPV6 client then an IPV4 client,
715 multiudpsink fell back to using the IPV6 socket.
716 When first connecting an IPV4 client, then an IPV6 client,
717 multiudpsink errored out, released the IPV4 socket, then
718 crashed when trying to send a message on NULL nevertheless,
719 that is however a separate issue.
720 This could probably be fixed by handling the setting of
721 sockets in multiudpsink after it has started, that will
722 however be a much more significant effort.
723 For now, this commit simply partially reverts the behaviour
724 of rtsp-stream: it will continue to only create the udpsinks
725 when needed, as was the case since the patches were merged,
726 it will however when creating them, always allocate both
727 sockets and set them on the sink before it starts, as was
728 the case prior to the patches.
729 Transport configuration will only error out if the allocation
730 of UDP sockets fails for the actual client's family, this
731 also downgrades the GST_ERRORs in alloc_ports_one_family
732 to GST_WARNINGs, as failing to allocate is no longer
734 https://bugzilla.gnome.org/show_bug.cgi?id=796875
736 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
740 meson: Convert common options to feature options
741 These are necessary for gst-build to set options correctly. The
742 remaining automagic option is cgroup support in examples.
743 https://bugzilla.gnome.org/show_bug.cgi?id=795107
745 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
747 * gst/rtsp-server/rtsp-stream.c:
748 rtsp-stream: Slightly simplify locking
750 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
752 * gst/rtsp-server/rtsp-client.c:
753 * gst/rtsp-server/rtsp-stream-transport.c:
754 * gst/rtsp-server/rtsp-stream-transport.h:
755 * gst/rtsp-server/rtsp-stream.c:
756 Limit queued TCP data messages to one per stream
757 Before, the watch backlog size in GstRTSPClient was changed
758 dynamically between unlimited and a fixed size, trying to avoid both
759 unlimited memory usage and deadlocks while waiting for place in the
760 queue. (Some of the deadlocks were described in a long comment in
762 In the previous commit, we changed to a fixed backlog size of 100.
763 This is possible, because we now handle RTP/RTCP data messages differently
764 from RTSP request/response messages.
765 The data messages are messages tunneled over TCP. We allow at most one
766 queued data message per stream in GstRTSPClient at a time, and
767 successfully sent data messages are acked by sending a "message-sent"
768 callback from the GstStreamTransport. Until that ack comes, the
769 GstRTSPStream does not call pull_sample() on its appsink, and
770 therefore the streaming thread in the pipeline will not be blocked
771 inside GstRTSPClient, waiting for a place in the queue.
772 pull_sample() is called when we have both an ack and a "new-sample"
773 signal from the appsink. Then, we know there is a buffer to write.
774 RTSP request/response messages are not acked in the same way as data
775 messages. The rest of the 100 places in the queue are used for
776 them. If the queue becomes full of request/response messages, we
777 return an error and close the connection to the client.
778 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
780 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
782 * gst/rtsp-server/rtsp-client.c:
783 rtsp-client: Use fixed backlog size
784 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
785 Preparation for the next commit, which changes to a different way of
786 avoiding both deadlocks and unlimited memory usage with the watch
789 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
791 * gst/rtsp-server/rtsp-media.c:
792 rtsp-media: unref clock (if set) when finalizing
793 https://bugzilla.gnome.org/show_bug.cgi?id=796814
795 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
797 * docs/libs/gst-rtsp-server-sections.txt:
798 rtsp-media: add gst_rtsp_media_*_set_clock to docs
799 https://bugzilla.gnome.org/show_bug.cgi?id=796814
801 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
803 * gst/rtsp-server/rtsp-media-factory.c:
804 media-factory: unref old clock when setting new clock
805 https://bugzilla.gnome.org/show_bug.cgi?id=796724
807 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
809 * gst/rtsp-server/rtsp-media-factory.c:
810 media-factory: unref clock in finalize
811 https://bugzilla.gnome.org/show_bug.cgi?id=796724
813 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
815 * gst/rtsp-server/rtsp-onvif-media.c:
816 rtsp-onvif-media: fix g-ir-scanner warnings
818 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
821 .gitignore: add another example binary
823 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
825 * examples/meson.build:
826 meson: add new test-appsrc2 example to meson build
828 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
830 * examples/Makefile.am:
831 examples: fix build of new test-appsrc2 example
832 Need to link against libgstapp-1.0.
834 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
836 * examples/.gitignore:
837 * examples/Makefile.am:
838 * examples/test-appsrc2.c:
839 examples: Add test-appsrc2
840 Add an example of feeding both audio and video into an RTSP
843 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
845 * gst/rtsp-server/rtsp-client.c:
846 client: Strip transport parts as whitespaces could be around commas
847 https://bugzilla.gnome.org/show_bug.cgi?id=758428
849 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
851 * gst/rtsp-server/rtsp-stream.c:
852 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
853 Fix race when setting up source elements.
854 Since we set the source element(s) to PLAYING state before hooking
855 them up to the downstream funnel, it's possible for the source element
856 to receive packets before we actually get to linking it to the funnel,
857 in which case buffers would be pushed out on an unlinked pad, causing
858 it to error out and stop receiving more data.
859 We fix this by blocking the source's srcpad until we have linked it.
860 https://bugzilla.gnome.org/show_bug.cgi?id=796160
862 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
864 * gst/rtsp-server/rtsp-stream.c:
865 rtsp-stream: Fix mismatch between allowed and configured protocols
866 https://bugzilla.gnome.org/show_bug.cgi?id=796679
868 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
870 * gst/rtsp-server/rtsp-stream.c:
871 rtsp-stream: Emit a signal when the SRTP decoder is created
872 https://bugzilla.gnome.org/show_bug.cgi?id=778080
874 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
876 * gst/rtsp-server/rtsp-stream.c:
877 rtsp-stream: Don't require presence of sinks in _get_*_socket()
878 Transport specific sink elements are added to the pipeline
879 in PLAY request and sockets are already created in SETUP so
880 it's actually wrong to require the presence of sinks in
881 _get_*_socket() functions.
882 https://bugzilla.gnome.org/show_bug.cgi?id=793441
884 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
886 * gst/rtsp-server/rtsp-stream.c:
887 rtsp-stream: Update transport for multicast clients as well
888 If a multicast client requests different transport settings
889 than the existing one make sure that this new transport
890 configuruation is propagated to the multicast udp sink.
891 https://bugzilla.gnome.org/show_bug.cgi?id=793441
893 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
895 * gst/rtsp-server/rtsp-stream.c:
896 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
897 And not on unicast udp sinks
898 https://bugzilla.gnome.org/show_bug.cgi?id=793441
900 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
902 * gst/rtsp-server/rtsp-address-pool.c:
903 * gst/rtsp-server/rtsp-auth.c:
904 * gst/rtsp-server/rtsp-client.c:
905 * gst/rtsp-server/rtsp-media-factory-uri.c:
906 * gst/rtsp-server/rtsp-media-factory.c:
907 * gst/rtsp-server/rtsp-media.c:
908 * gst/rtsp-server/rtsp-mount-points.c:
909 * gst/rtsp-server/rtsp-server.c:
910 * gst/rtsp-server/rtsp-session-media.c:
911 * gst/rtsp-server/rtsp-session-pool.c:
912 * gst/rtsp-server/rtsp-session.c:
913 * gst/rtsp-server/rtsp-stream-transport.c:
914 * gst/rtsp-server/rtsp-stream.c:
915 * gst/rtsp-server/rtsp-thread-pool.c:
916 Update for g_type_class_add_private() deprecation in recent GLib
918 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
920 * gst/rtsp-server/rtsp-auth.c:
921 * gst/rtsp-server/rtsp-media.c:
922 * gst/rtsp-server/rtsp-sdp.c:
923 * gst/rtsp-server/rtsp-stream.c:
926 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
928 * examples/Makefile.am:
929 * examples/test-video-disconnect.c:
930 examples: Add test-video-disconnect example
931 Simple example which cuts off all clients 10 seconds
932 after the first one connects.
934 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
936 * docs/libs/gst-rtsp-server-sections.txt:
937 * examples/test-auth-digest.c:
938 * gst/rtsp-server/rtsp-auth.c:
939 * gst/rtsp-server/rtsp-auth.h:
940 rtsp-auth: Add support for parsing .htdigest files
941 Passwords are usually not stored in clear text, but instead
942 stored already hashed in a .htdigest file.
943 Add support for parsing such files, add API to allow setting
944 a custom realm in RTSPAuth, and update the digest example.
945 https://bugzilla.gnome.org/show_bug.cgi?id=796637
947 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
949 * gst/rtsp-sink/gstrtspclientsink.c:
950 * gst/rtsp-sink/gstrtspclientsink.h:
951 rtspclientsink: fix waiting for multiple streams
952 We were previously only ever waiting for a single stream to notify it's
953 blocked status through GstRTSPStreamBlocking. Actually count streams to
955 Fixes rtspclientsink sending SDP's without out some of the input
957 https://bugzilla.gnome.org/show_bug.cgi?id=796624
959 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
961 * docs/libs/gst-rtsp-server-sections.txt:
962 docs: add missing auth methods
964 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
966 * gst/rtsp-server/rtsp-stream.c:
967 rtsp-stream: only create funnel if it didn't exist already.
968 This precented using multiple protocols for the same stream.
969 https://bugzilla.gnome.org/show_bug.cgi?id=796634
971 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
973 * examples/meson.build:
974 meson: build auth-digest example
976 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
978 * gst/rtsp-server/rtsp-client.c:
979 * gst/rtsp-server/rtsp-media.c:
980 * gst/rtsp-server/rtsp-sdp.c:
981 * gst/rtsp-server/rtsp-session-media.c:
982 * gst/rtsp-server/rtsp-stream-transport.c:
983 Get payloader stats only for the sending streams
984 Get/set payloader properties only for streams that actually
985 contain a payloader element.
986 https://bugzilla.gnome.org/show_bug.cgi?id=796523
988 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
990 * gst/rtsp-server/Makefile.am:
991 Makefile: Don't hardcode libtool for g-i build
992 Similar to the other commits in core/base/bad
994 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
996 * gst/rtsp-server/rtsp-onvif-media-factory.h:
997 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
998 https://bugzilla.gnome.org/show_bug.cgi?id=796229
1000 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
1002 * gst/rtsp-sink/gstrtspclientsink.c:
1003 rtspclientsink: Don't deadlock in preroll on early close
1004 If the connection is closed very early, the flushing
1005 marker might not get set and rtspclientsink can get
1006 deadlocked waiting for preroll forever.
1007 https://bugzilla.gnome.org/show_bug.cgi?id=786961
1009 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1012 * meson_options.txt:
1013 meson: Update option names to omit disable_ and with- prefixes
1014 Also yield common options to the outer project (gst-build in our case)
1015 so that they don't have to be set manually.
1017 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
1020 meson: use -Wl,-Bsymbolic-functions where supported
1021 Just like the autotools build.
1023 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1026 * tests/check/Makefile.am:
1027 configure: check for -good and -bad plugins only in uninstalled setup
1028 Avoids confusing configure messages looking or a -good .pc file
1030 Also use plugindir variables that common macros set while at it.
1031 https://bugzilla.gnome.org/show_bug.cgi?id=795466
1033 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
1035 * gst/rtsp-server/rtsp-client.c:
1036 rtsp-client: Fix session timeout
1037 When streaming data over TCP then is not the keep-alive
1038 functionality working.
1039 The reason is that the function do_send_data have changed
1040 to boolean but the code is still checking the received result
1041 from send_func with GST_RTSP_OK.
1042 The result is that a successful send_func will always lead to
1043 that do_send_data is returning false and the keep-alive will
1045 https://bugzilla.gnome.org/show_bug.cgi?id=795321
1047 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1049 * docs/libs/gst-rtsp-server-sections.txt:
1050 * gst/rtsp-server/rtsp-media.c:
1051 * gst/rtsp-server/rtsp-sdp.c:
1052 * gst/rtsp-server/rtsp-stream.c:
1053 * gst/rtsp-server/rtsp-stream.h:
1054 * gst/rtsp-sink/gstrtspclientsink.c:
1055 * gst/rtsp-sink/gstrtspclientsink.h:
1056 Implement support for ULP Forward Error Correction
1057 In this initial commit, interface is only exposed for RECORD,
1058 further work will be needed in rtspsrc to support this for
1060 https://bugzilla.gnome.org/show_bug.cgi?id=794911
1062 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1064 * gst/rtsp-server/rtsp-onvif-media.c:
1065 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
1066 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
1067 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
1068 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
1069 the opposite, just like the ONVIF standard.
1070 Let's follow those RFCs as we're doing RTSP here, and add a property at
1071 a later time if needed to switch to the SDP RFC behaviour.
1072 https://bugzilla.gnome.org/show_bug.cgi?id=793964
1074 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1077 Automatic update of common submodule
1078 From 3fa2c9e to ed78bee
1080 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
1082 * gst/rtsp-server/rtsp-client.c:
1083 * gst/rtsp-server/rtsp-media-factory.c:
1084 * gst/rtsp-server/rtsp-media.c:
1085 * gst/rtsp-server/rtsp-stream.c:
1086 * tests/check/gst/rtspclientsink.c:
1087 gst: Run everything through gst-indent again
1089 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
1091 * gst/rtsp-server/rtsp-media.c:
1092 * tests/check/gst/media.c:
1093 rtsp-media: query the position on active streams if media is complete
1094 If the media is complete, i.e. one or more streams have been configured
1095 with sinks, then we want to query the position on those streams only.
1096 A query on an incomplete stream may return a position that originates from
1098 https://bugzilla.gnome.org/show_bug.cgi?id=794964
1100 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1102 * gst/rtsp-sink/gstrtspclientsink.c:
1103 rtspclientsink: make sure not to use freed string
1104 Set transport string to NULL after freeing it, so that
1105 at worst we get a NULL pointer if constructing a new
1106 transport string fails (which shouldn't really fail here).
1107 Also check return value of that, just in case.
1110 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1112 * gst/rtsp-server/rtsp-client.c:
1113 rtsp-client: do not free string passed to take_header
1115 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1117 * gst/rtsp-server/rtsp-stream.c:
1118 rtsp-stream: do not take lock in request_aux_receiver
1119 Added it right before pushing the previous commit, it is
1120 incorrect and deadlocks because this function gets called
1121 from the join_bin thread, which already holds the lock,
1122 that's the reason why request_aux_sender didn't take the
1125 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1127 * docs/libs/gst-rtsp-server-sections.txt:
1128 * gst/rtsp-server/rtsp-media-factory.c:
1129 * gst/rtsp-server/rtsp-media-factory.h:
1130 * gst/rtsp-server/rtsp-media.c:
1131 * gst/rtsp-server/rtsp-media.h:
1132 * gst/rtsp-server/rtsp-stream.c:
1133 * gst/rtsp-server/rtsp-stream.h:
1134 rtsp-server: add API to enable retransmission requests
1135 "do-retransmission" was previously set when rtx-time != 0,
1136 which made no sense as do-retransmission is used to enable
1137 the sending of retransmission requests, where as rtx-time
1138 is used by the peer to enable storing of buffers in order
1139 to respond to retransmission requests.
1140 rtsp-media now also provides a callback for the
1141 request-aux-receiver signal.
1142 https://bugzilla.gnome.org/show_bug.cgi?id=794822
1144 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1146 * gst/rtsp-sink/gstrtspclientsink.c:
1147 rtspclientsink: add rtx ssrc to mikey's crypto sessions
1148 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1150 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1152 * gst/rtsp-sink/gstrtspclientsink.c:
1153 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
1154 This in order to be able to decrypt the RTCP backchannel
1155 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1157 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1159 * gst/rtsp-server/rtsp-client.c:
1160 rtsp-client: Send KeyMgmt header in ANNOUNCE response
1161 When sending back an encrypted RTCP back channel, it is useful
1162 for the client to know the encryption key.
1163 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1165 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1167 * gst/rtsp-server/rtsp-client.c:
1168 * gst/rtsp-server/rtsp-stream.c:
1169 * gst/rtsp-server/rtsp-stream.h:
1170 rtsp-stream: extract handle_keymgmt from rtsp-client
1171 rtspclientsink will also need to parse KeyMgmt headers
1172 sent by the server to decrypt the RTCP backchannel stream
1173 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1175 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1177 * gst/rtsp-sink/gstrtspclientsink.c:
1178 * tests/check/gst/rtspclientsink.c:
1179 rtspclientsink: Fix client ports for the RTCP backchannel
1180 This was broken since the work for delayed transport creation
1181 was merged: the creation of the transports string depends on
1182 calling stream_get_server_port, which only starts returning
1183 something meaningful after a call to stream_allocate_udp_sockets
1184 has been made, this function expects a transport that we parse
1185 from the transport string ...
1186 Significant refactoring is in order, but does not look entirely
1187 trivial, for now we put a band aid on and create a second transport
1188 string after the stream has been completed, to pass it in
1189 the request headers instead of the previous, incomplete one.
1190 https://bugzilla.gnome.org/show_bug.cgi?id=794789
1192 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
1194 * gst/rtsp-server/rtsp-client.c:
1195 rtsp-client:Error handling when equal http session cookie
1196 There are some clients that are sending same session cookie on random
1198 https://bugzilla.gnome.org/show_bug.cgi?id=753616
1200 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1202 * gst/rtsp-server/rtsp-media-factory-uri.c:
1203 rtsp-media-factory-uri: Fix compilation with latest GLib
1204 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
1205 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
1206 data->factory = g_object_ref (factory);
1209 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1217 === release 1.14.0 ===
1219 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1225 * gst-rtsp-server.doap:
1229 === release 1.13.91 ===
1231 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
1237 * gst-rtsp-server.doap:
1241 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1243 * gst/rtsp-server/Makefile.am:
1244 * gst/rtsp-server/meson.build:
1245 * gst/rtsp-server/rtsp-address-pool.h:
1246 * gst/rtsp-server/rtsp-auth.h:
1247 * gst/rtsp-server/rtsp-client.h:
1248 * gst/rtsp-server/rtsp-context.h:
1249 * gst/rtsp-server/rtsp-media-factory-uri.h:
1250 * gst/rtsp-server/rtsp-media-factory.h:
1251 * gst/rtsp-server/rtsp-media.h:
1252 * gst/rtsp-server/rtsp-mount-points.h:
1253 * gst/rtsp-server/rtsp-onvif-client.h:
1254 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1255 * gst/rtsp-server/rtsp-onvif-media.h:
1256 * gst/rtsp-server/rtsp-onvif-server.h:
1257 * gst/rtsp-server/rtsp-params.h:
1258 * gst/rtsp-server/rtsp-permissions.h:
1259 * gst/rtsp-server/rtsp-sdp.h:
1260 * gst/rtsp-server/rtsp-server-prelude.h:
1261 * gst/rtsp-server/rtsp-server.h:
1262 * gst/rtsp-server/rtsp-session-media.h:
1263 * gst/rtsp-server/rtsp-session-pool.h:
1264 * gst/rtsp-server/rtsp-session.h:
1265 * gst/rtsp-server/rtsp-stream-transport.h:
1266 * gst/rtsp-server/rtsp-stream.h:
1267 * gst/rtsp-server/rtsp-thread-pool.h:
1268 * gst/rtsp-server/rtsp-token.h:
1269 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
1270 We need different export decorators for the different libs.
1271 For now no actual change though, just rename before the release,
1272 and add prelude headers to define the new decorator to GST_EXPORT.
1274 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1276 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1277 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
1278 https://bugzilla.gnome.org/show_bug.cgi?id=794143
1280 === release 1.13.90 ===
1282 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
1288 * gst-rtsp-server.doap:
1292 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1294 * gst/rtsp-server/rtsp-media-factory.c:
1295 * gst/rtsp-server/rtsp-permissions.c:
1296 permissions: add Since tags and example for new API
1298 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1300 * docs/libs/gst-rtsp-server-sections.txt:
1301 * gst/rtsp-server/rtsp-media-factory.c:
1302 * gst/rtsp-server/rtsp-media-factory.h:
1303 * gst/rtsp-server/rtsp-permissions.c:
1304 * gst/rtsp-server/rtsp-permissions.h:
1305 * tests/check/gst/permissions.c:
1306 permissions: more bindings-friendly API
1307 https://bugzilla.gnome.org/show_bug.cgi?id=793975
1309 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1312 meson: enable more warnings
1314 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1316 * gst/rtsp-server/rtsp-client.c:
1317 rtsp-client: Place netaddress meta on packets received via TCP
1318 This allows us to later map signals from rtpbin/rtpsource back to the
1319 corresponding stream transport, and allows to do keep-alive based on
1320 RTCP packets in case of TCP media transport.
1321 https://bugzilla.gnome.org/show_bug.cgi?id=789646
1323 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1325 * gst/rtsp-sink/gstrtspclientsink.c:
1326 rtspclientsink: if OPEN failed, unqueue next command
1327 As READY_TO_PAUSED can no longer return async, the RECORD
1328 command will be queued before the OPEN command fails
1329 (for example in case the server could not be connected),
1330 and record then waits for ever.
1331 https://bugzilla.gnome.org/show_bug.cgi?id=793896
1333 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1335 * gst/rtsp-sink/gstrtspclientsink.c:
1336 rtspclientsink: fix retrieval of custom payloader caps
1337 If a bin is passed as the custom payloader, the caps of
1338 its factory will be empty, the correct way to obtain the caps
1339 is to query its sinkpad.
1341 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1343 * gst/rtsp-sink/gstrtspclientsink.c:
1344 rtspclientsink: fix extra unref of custom payloader
1346 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1348 * gst/rtsp-sink/gstrtspclientsink.c:
1349 rspclientsink: fix recent code indentation
1351 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1353 * gst/rtsp-sink/gstrtspclientsink.c:
1354 rtspclientsink: add missing get_type prototype
1356 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1358 * gst/rtsp-sink/gstrtspclientsink.c:
1359 rtspclientsink: allow setting payloader as pad property
1360 This was a FIXME item, and can be quite useful, also
1361 allowing to specify payloader properties from the command
1362 line, which is always nice.
1363 https://bugzilla.gnome.org/show_bug.cgi?id=793776
1365 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
1367 * gst/rtsp-server/rtsp-media.c:
1368 rtsp-media: Replace g_print() log line
1369 https://bugzilla.gnome.org/show_bug.cgi?id=793838
1371 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1373 * gst/rtsp-server/rtsp-media.c:
1374 * tests/check/gst/rtspclientsink.c:
1375 rtsp-media: fix RECORD getting stuck
1376 The test_record case was working because async=false had
1377 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
1378 but that was incorrect, as it should not be needed.
1379 Removing async=false made the test fail as expected, this is
1380 fixed by not trying to preroll when preparing the media for
1381 RECORD, as start_prepare is called upon receiving ANNOUNCE,
1382 and our peer will not start sending media until it has received
1383 a response to that request, and sent and received a response
1384 to RECORD as well, thus obviously preventing preroll.
1385 https://bugzilla.gnome.org/show_bug.cgi?id=793738
1387 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1389 * gst/rtsp-server/rtsp-auth.c:
1390 rtsp-auth: fix set_tls_authentication_mode annotation
1392 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
1394 * gst/rtsp-server/rtsp-onvif-media.c:
1395 rtp-server: remove redefined variable
1396 res is a boolean variable which is defined in the function scope and
1397 redefined, with no reason, in the loop scope. This patch removes the
1399 https://bugzilla.gnome.org/show_bug.cgi?id=793592
1401 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
1403 * gst/rtsp-server/rtsp-media.c:
1404 * gst/rtsp-server/rtsp-stream.c:
1405 * gst/rtsp-server/rtsp-stream.h:
1406 stream: Add functions for checking if stream is receiver or sender
1407 ...and replace all checks for RECORD in GstRTSPMedia which are really
1408 for "sender-only". This way the code becomes more generic and introducing
1409 support for onvif-backchannel later on will require no changes in
1412 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
1414 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1415 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1416 onvif: Make requires_backchannel() public
1417 ...in order to let subclasses building the onvif part of the pipeline
1418 check whether backchannel shall be included or not.
1420 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1422 * gst/rtsp-server/rtsp-onvif-media.c:
1423 rtsp-server: Switch around sendonly/recvonly attributes
1424 They are wrong in the ONVIF streaming spec. The backchannel should be
1425 recvonly and the normal media should be sendonly: direction is always
1426 from the point of view of the SDP offerer (the server) according to
1429 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1431 * docs/libs/gst-rtsp-server-docs.sgml:
1432 * docs/libs/gst-rtsp-server-sections.txt:
1433 * examples/.gitignore:
1434 * examples/Makefile.am:
1435 * examples/test-onvif-backchannel.c:
1436 * gst/rtsp-server/Makefile.am:
1437 * gst/rtsp-server/rtsp-media.h:
1438 * gst/rtsp-server/rtsp-onvif-client.c:
1439 * gst/rtsp-server/rtsp-onvif-client.h:
1440 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1441 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1442 * gst/rtsp-server/rtsp-onvif-media.c:
1443 * gst/rtsp-server/rtsp-onvif-media.h:
1444 * gst/rtsp-server/rtsp-onvif-server.c:
1445 * gst/rtsp-server/rtsp-onvif-server.h:
1446 * gst/rtsp-server/rtsp-sdp.c:
1447 * gst/rtsp-server/rtsp-sdp.h:
1448 rtsp: Add support for ONVIF backchannel
1449 This adds a new RTSP server, client, media-factory and media subclass
1450 for handling the specifics of the backchannel. Ideally this later can be
1451 extended with other ONVIF specific features.
1453 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1455 * gst/rtsp-server/rtsp-media.c:
1456 rtsp-media: Add support for sending+receiving medias
1457 We need to add an appsrc/appsink in that case because otherwise the
1458 media bin will be a sink and a source for rtpbin, causing a pipeline
1460 https://bugzilla.gnome.org/show_bug.cgi?id=788950
1462 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1468 === release 1.13.1 ===
1470 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1474 * gst-rtsp-server.doap:
1478 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1480 * gst/rtsp-server/rtsp-session-pool.c:
1481 session-pool: remove nullable return annotation
1482 create_watch can only return NULL from the API guards, no
1485 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1487 * gst/rtsp-server/rtsp-media-factory.c:
1488 * gst/rtsp-server/rtsp-media.c:
1489 set_clock functions: Add nullable annotations
1491 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1493 * gst/rtsp-server/rtsp-auth.c:
1494 * gst/rtsp-server/rtsp-client.c:
1495 * gst/rtsp-server/rtsp-media-factory.c:
1496 * gst/rtsp-server/rtsp-media.c:
1497 * gst/rtsp-server/rtsp-mount-points.c:
1498 * gst/rtsp-server/rtsp-server.c:
1499 * gst/rtsp-server/rtsp-session-media.c:
1500 * gst/rtsp-server/rtsp-session-pool.c:
1501 * gst/rtsp-server/rtsp-session.c:
1502 * gst/rtsp-server/rtsp-stream-transport.c:
1503 * gst/rtsp-server/rtsp-stream.c:
1504 * gst/rtsp-server/rtsp-thread-pool.c:
1505 All around: add annotations and API guards
1507 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1509 * tests/test-cleanup.c:
1510 test-cleanup: bind any port
1511 The meson test suite runs tests in parallel, trying to bind
1512 a single port made the test fail.
1514 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1517 meson: make version numbers ints and fix int/string comparison
1518 WARNING: Trying to compare values of different types (str, int).
1519 The result of this is undefined and will become a hard error
1520 in a future Meson release.
1522 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1524 * gst/rtsp-server/rtsp-context.c:
1525 gst_rtsp_context_get_current: add (skip) annotation
1526 The return value type is defined with G_DEFINE_POINTER_TYPE,
1527 and gi emits the following warning:
1528 Invalid non-constant return of bare structure or union; register as
1529 boxed type or (skip)
1531 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1533 * gst/rtsp-server/rtsp-client.c:
1534 rtsp-client: add type annotations
1535 gi doesn't seem to be able to figure out the type of the
1536 signal parameters when defined with G_DEFINE_POINTER_TYPE
1538 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
1541 autotools: use -fno-strict-aliasing where supported
1542 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1544 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1547 meson: use -fno-strict-aliasing where supported
1548 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1550 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1552 * gst/rtsp-server/rtsp-mount-points.c:
1553 mount-points: bail out of loop again when matching mount points
1554 Previous patch led to us iterating the entire sequence. Bail out
1555 of the loop again if we have a match but are moving away from it.
1556 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1558 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1560 * tests/check/gst/mountpoints.c:
1561 tests: mountpoints: add more checks for mount point path matching
1562 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1564 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
1566 * gst/rtsp-server/rtsp-mount-points.c:
1567 mount-points: fix matching of paths where there's also an entry with a common prefix
1568 e.g. with the following mount points
1572 _match() would not match /raw/video and /raw/snapshot correctly.
1573 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1575 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1577 * docs/libs/gst-rtsp-server-sections.txt:
1578 * gst/rtsp-server/rtsp-permissions.c:
1579 * gst/rtsp-server/rtsp-permissions.h:
1580 * tests/check/gst/permissions.c:
1581 permissions: add some new API to make this usable from bindings
1582 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1584 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
1586 * gst/rtsp-server/rtsp-token.c:
1587 rtsp-token: annotate constructors for bindings
1588 This maps _new_empty() to _new(), which also makes RTSPToken()
1589 work properly now. Since this API wasn't usable from bindings
1590 before, this should hopefully be fine.
1591 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1593 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1595 * docs/libs/gst-rtsp-server-sections.txt:
1596 * gst/rtsp-server/rtsp-token.c:
1597 * gst/rtsp-server/rtsp-token.h:
1598 * tests/check/gst/token.c:
1599 rtsp-token: add some API to set fields from bindings
1600 The existing functions are all vararg-based and as such
1601 not usable from bindings.
1602 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1604 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1606 * tests/check/gst/rtspclientsink.c:
1607 * tests/check/gst/rtspserver.c:
1608 * tests/check/gst/sessionpool.c:
1609 * tests/check/gst/stream.c:
1610 tests: fix indentation
1613 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
1615 * tests/check/gst/rtspserver.c:
1616 tests: rtspserver: fix another ref leak
1617 Even if this didn't show up in valgrind.
1619 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1621 * tests/check/gst/rtspclientsink.c:
1622 tests: rtspclientsink: fix leak
1624 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
1626 * tests/check/gst/rtspserver.c:
1627 test: rtspserver: plug memory leak in test_no_session_timeout
1628 In test_no_session_timeout, unref the rtsp session object when the
1630 https://bugzilla.gnome.org/show_bug.cgi?id=792127
1632 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
1634 * gst/rtsp-sink/gstrtspclientsink.c:
1635 rtpsclientsink: Initialize and clear newly added mutex and cond
1636 While it *did* work, glib would automatically create new mutex and cond
1637 ... which never got freed
1639 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1641 * gst/rtsp-server/rtsp-stream.c:
1642 rtsp-stream: Set multicast TTL on the multicast sockets
1643 And not if we do unicast UDP.
1644 https://bugzilla.gnome.org/show_bug.cgi?id=791743
1646 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
1648 * gst/rtsp-server/rtsp-stream.c:
1649 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
1650 In the multicast case (as in test-multicast, not test-multicast2), the
1651 address could be allocated/reserved (and thus set) already without
1652 allocating the actual socket. We need to allocate the socket here still
1653 instead of just claiming that it was already allocated.
1654 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
1656 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1658 * gst/rtsp-sink/gstrtspclientsink.c:
1659 * gst/rtsp-sink/gstrtspclientsink.h:
1660 rtspclientsink: Use the new rtsp-stream API
1661 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1663 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1665 * gst/rtsp-sink/gstrtspclientsink.c:
1666 * gst/rtsp-sink/gstrtspclientsink.h:
1667 rtspclientsink: Wait until OPEN has been scheduled
1668 Make sure that the sink thread has started opening connection
1669 to the server before continuing.
1670 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1672 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
1675 Automatic update of common submodule
1676 From e8c7a71 to 3fa2c9e
1678 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
1680 * gst/rtsp-server/rtsp-media.c:
1681 * gst/rtsp-server/rtsp-session-media.c:
1682 * gst/rtsp-server/rtsp-stream.c:
1683 rtsp-server: Minor doc fixes
1686 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1689 * tests/Makefile.am:
1690 tests: disable all tests when --disable-tests is used
1691 Move conditional subdir include into top level.
1692 Based on patch by: Joel Holdsworth
1693 https://bugzilla.gnome.org/show_bug.cgi?id=757703
1695 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
1698 * meson_options.txt:
1699 * tests/meson.build:
1700 meson: build more tests and add options to disable tests and examples
1702 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
1704 * gst/rtsp-server/rtsp-session.c:
1705 Fix build when -Werror=deprecated-declarations is on
1706 As gst_rtsp_session_next_timeout is deprecated.
1708 ../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]
1709 res = (gst_rtsp_session_next_timeout (session, now) == 0);
1711 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
1712 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
1713 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
1716 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
1719 Automatic update of common submodule
1720 From 3f4aa96 to e8c7a71
1722 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1724 * tests/check/gst/media.c:
1725 check/media: Add seekability test case: not all streams are active
1726 Media contains two streams but only one is complete and prepared
1728 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1730 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1732 * gst/rtsp-server/rtsp-stream.c:
1733 rtsp-stream: Do not reset 'blocking' if stream is already blocked
1734 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1736 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1738 * gst/rtsp-server/rtsp-media.c:
1739 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
1740 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1742 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
1745 meson: remove vs_module_defs_dir variable which is no longer needed
1747 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
1749 * gst/rtsp-server/rtsp-session.h:
1752 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
1755 * gst/rtsp-server/meson.build:
1757 * win32/common/libgstrtspserver.def:
1758 win32: remove .def file with exports
1759 They're no longer needed, symbol exporting is now explicit
1760 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
1762 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1765 autotools: stop controlling symbol visibility with -export-symbols-regex
1766 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
1767 This should result in consistent behaviour for the autotools and
1770 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1772 * gst/rtsp-server/rtsp-media.h:
1773 * gst/rtsp-server/rtsp-server.h:
1774 * gst/rtsp-server/rtsp-session.c:
1775 * gst/rtsp-server/rtsp-session.h:
1776 rtsp-server: add missing GST_EXPORT and export deprecated funcs
1778 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
1780 * tests/check/gst/media.c:
1781 check: Add seekability testing on medias
1782 Make sure that once GstRTSPMedia are prepared they returned
1783 the expected seekability results
1784 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1786 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
1788 * docs/libs/gst-rtsp-server-sections.txt:
1789 * gst/rtsp-server/rtsp-media.c:
1790 * gst/rtsp-server/rtsp-stream.c:
1791 * gst/rtsp-server/rtsp-stream.h:
1792 * win32/common/libgstrtspserver.def:
1793 rtsp-media: Enable seeking query before pipeline is complete
1794 SDP are now provided *before* the pipeline is fully complete. In order
1795 to know whether a media is seekable or not therefore requires asking
1796 the invididual streams.
1797 API: gst_rtsp_stream_seekable
1798 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1800 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
1802 * gst/rtsp-server/rtsp-media.c:
1803 rtsp-media: Fix handling in default_unsuspend()
1804 Handle the case when streams are not blocked and media
1805 is suspended from PAUSED.
1806 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
1807 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1809 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
1811 * tests/check/gst/media.c:
1812 check/media: Fix thread pool leak.
1813 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
1814 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1816 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
1818 * gst/rtsp-server/rtsp-media.c:
1819 rtsp-media: Removed fakesink elements
1820 There is not need of adding fakesink elements to the media
1821 pipeline in the dynamic-payloader case.
1822 The media pipeline itself is dynamically updated with
1823 the receiver and sender parts that are based on the client
1824 transport information known after SETUP has been received.
1825 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
1826 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1828 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
1830 * gst/rtsp-server/rtsp-media.c:
1831 rtsp-media: Corrected ASYNC_DONE handling
1832 Media is complete when all the transport based parts are
1833 added to the media pipeline. At this point ASYNC_DONE is
1834 posted by the media pipeline and media is ready to enter
1836 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
1837 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1839 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
1841 * tests/check/gst/media.c:
1842 check/media: Check that prepared media can provide a SDP
1843 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
1845 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
1847 * gst/rtsp-server/rtsp-client.c:
1848 rtsp-client: Don't leak addr
1851 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
1853 * gst/rtsp-server/rtsp-client.c:
1854 * gst/rtsp-server/rtsp-session-media.c:
1855 * gst/rtsp-server/rtsp-stream.c:
1858 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
1860 * gst/rtsp-server/rtsp-media.c:
1861 rtsp-media: Don't unblock with remaining dynamic payloaders
1862 If we still have some dynamic paylaoders which haven't posted
1863 no-more-pads yet, don't go to PREPARED if one of the streams
1865 The risk was that we would end up not exposing/using all specified
1867 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
1868 then it will take a bit more time to start. But only if those 3
1869 conditions are present.
1870 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1872 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
1874 * gst/rtsp-server/rtsp-media.c:
1877 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
1879 * gst/rtsp-server/rtsp-media.c:
1880 rtsp-media: Don't set float on a gint64 variable
1881 Just use 0. Fixes 'undefined' behaviour from clang
1883 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
1885 * gst/rtsp-server/rtsp-media.c:
1886 rtsp-media: Fix previous commit
1887 We only want to count dynamic payloaders
1889 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
1891 * gst/rtsp-server/rtsp-media.c:
1892 * tests/check/gst/media.c:
1893 rtsp-media: Handle multiple dynamic elements
1894 If we have more than one dynamic payloader in the pipeline, we need
1895 to wait until the *last* one emits 'no-more-pads' before switching
1897 Failure to do so would result in a race where some of the streams
1898 wouldn't properly be prepared
1899 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1901 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1903 * win32/common/libgstrtspserver.def:
1904 win32: Fix exported symbols list
1906 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1908 * gst/rtsp-server/rtsp-stream.c:
1909 rtsp-stream: Only update the RTP udpsink if it actually exists
1910 For send-only streams it does not exist, but the RTCP udpsink might.
1912 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
1914 * win32/common/libgstrtspserver.def:
1915 win32: Update exports
1917 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
1919 * gst/rtsp-server/rtsp-media.c:
1920 * gst/rtsp-server/rtsp-stream.c:
1921 * gst/rtsp-server/rtsp-stream.h:
1922 rtsp-media: seek on media pipelines that are complete
1923 Make sure that a seek is performed on pipelines that
1924 contain at least one sink element.
1925 Change-Id: Icf398e10add3191d104b1289de612412da326819
1926 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1928 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
1930 * gst/rtsp-server/rtsp-client.c:
1931 * gst/rtsp-server/rtsp-media.c:
1932 * gst/rtsp-server/rtsp-media.h:
1933 * gst/rtsp-server/rtsp-stream.c:
1934 * gst/rtsp-server/rtsp-stream.h:
1935 * tests/check/gst/client.c:
1936 * tests/check/gst/media.c:
1937 * tests/check/gst/rtspserver.c:
1938 * tests/check/gst/stream.c:
1939 Dynamically reconfigure pipeline in PLAY based on transports
1940 The initial pipeline does not contain specific transport
1941 elements. The receiver and the sender parts are added
1943 If the media is shared, the streams are dynamically
1944 reconfigured after each PLAY.
1945 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1947 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
1949 * gst/rtsp-server/rtsp-stream.c:
1950 rtsp-stream: obtain stream position from pad
1951 If no sinks have been added yet, obtain the current and
1952 the stop position of the stream from the send_src pad.
1953 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
1954 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1956 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
1958 * gst/rtsp-server/rtsp-session-media.c:
1959 * gst/rtsp-server/rtsp-session-media.h:
1960 rtsp-session-media: add function to get a list of transports
1961 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
1962 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1964 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
1966 * gst/rtsp-server/rtsp-stream.c:
1967 * gst/rtsp-server/rtsp-stream.h:
1968 rtsp-stream: add functions to get rtp and rtcp multicast sockets
1969 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
1970 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1972 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
1974 * gst/rtsp-server/rtsp-stream.c:
1975 stream: set async=sync=false only for RTCP appsink
1976 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
1977 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1979 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
1981 * gst/rtsp-server/rtsp-media.c:
1982 rtsp-media: return minimum value in query position case
1983 The minimum position should be returned as we are interested
1984 in the whole interval.
1985 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
1986 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1988 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
1990 * gst/rtsp-server/rtsp-session.c:
1991 * tests/check/gst/rtspserver.c:
1992 rtsp-session: Handle the case when timeout=0
1993 According to the documentation, a timeout of value 0 means
1994 that the session never timeouts. This adds handling of that.
1995 If timeout=0 we just return with a -1 from
1996 gst_rtsp_session_next_timeout_usec ().
1997 https://bugzilla.gnome.org/show_bug.cgi?id=785058
1999 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2001 * gst/rtsp-sink/gstrtspclientsink.c:
2002 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
2003 https://bugzilla.gnome.org/show_bug.cgi?id=785024
2005 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2007 * docs/libs/gst-rtsp-server-sections.txt:
2008 * gst/rtsp-server/rtsp-media-factory.c:
2009 docs: add media factory transport mode accessors
2010 and fix the documentation for the return value of the getter
2012 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
2014 * gst/rtsp-server/rtsp-client.c:
2015 rtsp-client: unref 'pipelined_requests' in finalize
2016 The hash table priv->pipelined_requests is not unref:ed in the
2017 finalize funktion. Make sure it is.
2018 https://bugzilla.gnome.org/show_bug.cgi?id=788704
2020 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
2022 * gst/rtsp-server/rtsp-media.c:
2023 rtsp-media: Initialize scalar variable
2026 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
2028 * win32/common/libgstrtspserver.def:
2029 win32: Update export file
2031 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2033 * gst/rtsp-server/rtsp-client.c:
2034 * gst/rtsp-server/rtsp-media.c:
2035 * gst/rtsp-server/rtsp-media.h:
2036 Start support for RTSP 2.0
2037 This adds basic support for new 2.0 features, though the protocol is
2038 subposdely backward incompatible, most semantics are the sames.
2041 * version negotiation
2042 * pipelined requests support
2043 * Media-Properties support
2044 * Accept-Ranges support
2046 * gst_rtsp_media_seekable
2047 The RTSP methods that have been removed when using 2.0 now return
2049 https://bugzilla.gnome.org/show_bug.cgi?id=781446
2051 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2053 * gst/rtsp-server/rtsp-stream.c:
2054 stream: Use stream duration as stream-stop if segment was not configured with a stop
2055 Allowing client to know stream duration when no seeking happened.
2056 https://bugzilla.gnome.org/show_bug.cgi?id=783435
2058 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
2060 * gst/rtsp-server/rtsp-media-factory.c:
2061 rtsp-media-factory: Don't cache any media if NULL was returned as key
2062 The docs already mentioned this, but we actually stored it in the hash
2063 table with key==NULL and leaked its reference forever.
2065 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
2067 * gst/rtsp-sink/gstrtspclientsink.c:
2068 * gst/rtsp-sink/gstrtspclientsink.h:
2069 rtspclientsink: Use a mutex for protecting against concurrent send/receives
2070 This is a simple port of:
2071 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
2072 * c438545dc9e2f14f657bc0ef261fff726449867b
2073 * cd17c71dcea5c9310d21f1347c7520983e5869ac
2074 in gst-plugins-good.
2076 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
2078 * gst/rtsp-server/rtsp-sdp.c:
2079 sdp: fix Memory leak in error case
2080 https://bugzilla.gnome.org/show_bug.cgi?id=787059
2082 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2084 * pkgconfig/meson.build:
2085 meson: don't install -uninstalled.pc file
2086 https://bugzilla.gnome.org/show_bug.cgi?id=786457
2088 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
2091 Automatic update of common submodule
2092 From 48a5d85 to 3f4aa96
2094 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2096 * gst/rtsp-server/rtsp-client.c:
2097 rtsp-client: Fix typo in debug message
2099 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2102 meson: hide symbols by default unless explicitly exported
2104 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2106 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2107 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
2108 Fixes meson warning about undefined @srcdir@.
2110 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
2112 * tests/meson.build:
2113 meson: skip tests on windows for now
2114 As we do in the other modules. As libgstcheck is currently not
2115 built on windows. Fixes "Fallback variable 'gst_check_dep' in
2116 the subproject 'gstreamer' does not exist"" Meson error.
2118 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
2120 * gst/rtsp-server/rtsp-stream.c:
2121 rtsp-stream: fix connection delay due to wrong assumption on last-sample
2122 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
2123 multiudpsink's last-sample always comes from the payloader. Which
2124 is wrong if auxiliary streams are multiplexed in the same stream.
2125 So check the buffer's ssrc against the caps'ssrc before to use its
2126 seqnum. If not the same ssrc just use the payloader as done prior
2127 the commit above or when there is no last-sample yet.
2128 https://bugzilla.gnome.org/show_bug.cgi?id=784094
2130 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2133 meson: Allow using glib as a subproject
2135 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2138 meson: fix with-package-name option
2139 https://bugzilla.gnome.org/show_bug.cgi?id=784082
2141 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2144 Distribute meson_options.txt
2146 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2149 And config.h.meson is no longer dist either
2151 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
2155 meson: config.h.meson is no longer needed
2157 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2159 * tests/check/meson.build:
2160 * tests/meson.build:
2161 meson: Fix building tests and activate them again
2163 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2165 * tests/check/meson.build:
2166 meson: Do not use path separator in test names
2167 Avoiding warnings like:
2168 WARNING: Target "elements/audioamplify" has a path separator in its name.
2170 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
2173 * meson_options.txt:
2174 meson: add options to set package name and origin
2175 https://bugzilla.gnome.org/show_bug.cgi?id=782172
2177 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2179 * gst/rtsp-server/rtsp-address-pool.h:
2180 * gst/rtsp-server/rtsp-auth.h:
2181 * gst/rtsp-server/rtsp-client.h:
2182 * gst/rtsp-server/rtsp-context.h:
2183 * gst/rtsp-server/rtsp-media-factory-uri.h:
2184 * gst/rtsp-server/rtsp-media-factory.h:
2185 * gst/rtsp-server/rtsp-media.h:
2186 * gst/rtsp-server/rtsp-mount-points.h:
2187 * gst/rtsp-server/rtsp-params.h:
2188 * gst/rtsp-server/rtsp-permissions.h:
2189 * gst/rtsp-server/rtsp-sdp.h:
2190 * gst/rtsp-server/rtsp-server.h:
2191 * gst/rtsp-server/rtsp-session-media.h:
2192 * gst/rtsp-server/rtsp-session-pool.h:
2193 * gst/rtsp-server/rtsp-session.h:
2194 * gst/rtsp-server/rtsp-stream-transport.h:
2195 * gst/rtsp-server/rtsp-stream.h:
2196 * gst/rtsp-server/rtsp-thread-pool.h:
2197 * gst/rtsp-server/rtsp-token.h:
2198 Mark symbols explicitly for export with GST_EXPORT
2200 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2203 * gst/rtsp-sink/Makefile.am:
2204 Remove plugin specific static build option
2205 Static and dynamic plugins now have the same interface. The standard
2206 --enable-static/--enable-shared toggle are sufficient.
2208 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2214 === release 1.12.0 ===
2216 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2222 * gst-rtsp-server.doap:
2226 === release 1.11.91 ===
2228 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
2234 * gst-rtsp-server.doap:
2238 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
2241 Automatic update of common submodule
2242 From 60aeef6 to 48a5d85
2244 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2246 * gst/rtsp-server/rtsp-media-factory.c:
2247 * gst/rtsp-server/rtsp-media.c:
2248 * gst/rtsp-server/rtsp-session.c:
2249 * gst/rtsp-server/rtsp-stream.c:
2250 gi: Fix some annotations and docstrings
2252 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2254 * gst/rtsp-server/meson.build:
2256 * meson_options.txt:
2259 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2263 Automatic update of common submodule
2264 From 39ac2f5 to 60aeef6
2266 === release 1.11.90 ===
2268 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
2274 * gst-rtsp-server.doap:
2278 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
2280 * examples/test-launch.c:
2281 examples: make test-launch pipeline shared by default as well
2283 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
2285 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2286 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
2287 Just the build dir is not going to work for srcdir!=builddir.
2289 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
2292 meson: Update version
2294 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
2299 === release 1.11.2 ===
2301 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2307 * gst-rtsp-server.doap:
2310 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
2313 meson: dist meson build files
2314 Ship meson build files in tarballs, so people who use tarballs
2315 in their builds can start playing with meson already.
2317 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
2319 * examples/test-record.c:
2320 examples/test-record: Add extra line to initial printout
2321 Add an example line of how to deliver a stream to the
2324 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2326 * gst/rtsp-server/rtsp-client.c:
2327 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
2328 If there is no Content-Length header, no body would be allocated and the
2329 '\0' would also not be appended to the body.
2331 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2333 * gst/rtsp-server/rtsp-client.c:
2334 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
2335 While they logically have 0 bytes length, GstRTSPConnection is appending
2336 a '\0' to everything making the size be 1 instead.
2338 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2343 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
2345 * gst/rtsp-server/rtsp-session.c:
2346 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
2347 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
2350 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
2355 === release 1.11.1 ===
2357 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2363 * gst-rtsp-server.doap:
2364 * win32/common/libgstrtspserver.def:
2367 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
2369 * gst/rtsp-server/rtsp-stream.c:
2370 rtsp-stream: corrected if-statement in _get_server_port()
2371 This bug was accidentally introduced while fixing a segfault
2372 in _get_server_port() function.
2373 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2375 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
2377 * gst/rtsp-server/rtsp-stream.c:
2378 * tests/check/gst/stream.c:
2379 rtsp-stream: fixed segmenation fault in _get_server_port()
2380 Calling function gst_rtsp_stream_get_server_port() results in
2381 segmenation fault in the RTP/RTSP/TCP case.
2382 Port that the server will use to receive RTCP makes only
2383 sense in the UDP case, however the function should handle
2384 the TCP case in a nicer way.
2385 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2387 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
2389 * gst/rtsp-server/rtsp-media-factory.c:
2390 dosc: Fix a little typo
2391 https://bugzilla.gnome.org/show_bug.cgi?id=777037
2393 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2395 * pkgconfig/Makefile.am:
2396 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2397 * pkgconfig/meson.build:
2398 meson: generate pkg-config -uninstalled pc files
2399 Generating those files is useful for users building the GStreamer stack
2400 using meson and having to link it to another project which is still
2401 using the autotools.
2402 https://bugzilla.gnome.org/show_bug.cgi?id=776810
2404 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2406 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2407 pkgconfig: fix -uninstalled pc file
2408 pcfiledir was never defined so the paths were wrong.
2409 https://bugzilla.gnome.org/show_bug.cgi?id=776867
2411 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
2413 * gst/rtsp-server/rtsp-stream.c:
2414 * tests/check/gst/rtspserver.c:
2415 rtsp-stream: Fixed TCP transport case
2416 Make sure that the appsink element is actually added to
2417 the bin before trying to link it with the elements in it.
2418 https://bugzilla.gnome.org/show_bug.cgi?id=776343
2420 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2426 Remove generated .spec file
2427 Likely extremely bitrotten, and we should not ship this anyway.
2429 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
2432 Automatic update of common submodule
2433 From f980fd9 to 39ac2f5
2435 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
2437 * gst/rtsp-server/rtsp-media.c:
2438 media: Fix pt map caps
2439 Since decryption is handled within rtpbin, all outcoming stream
2440 caps will be application/x-rtp (i.e. regular rtp)
2441 Fixes RECORD with SRTP streams
2443 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
2445 * gst/rtsp-server/rtsp-media-factory.c:
2446 media-factory: Create media objects with the proper transport mode
2447 The function called immediately afterwards (collect_streams()) will
2448 need it to work properly
2450 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
2452 * gst/rtsp-server/rtsp-auth.c:
2453 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2455 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
2457 * gst/rtsp-server/rtsp-media-factory.c:
2458 rtsp-media-factory: Don't create a pipeline for the media pipeline string
2459 We're going to put a pipeline into a pipeline otherwise, which is not
2462 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
2464 * gst/rtsp-server/rtsp-media.c:
2465 media: Fix race condition around finish_unprepare() if called multiple time
2466 https://bugzilla.gnome.org/show_bug.cgi?id=755329
2468 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
2470 * gst/rtsp-sink/gstrtspclientsink.c:
2471 rtspclientsink: Don't leave stale pointer after unref
2472 Fix a warning on shutdown - don't keep a pointer to an
2473 alread-unreffed object.
2475 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2478 common: use https protocol for common submodule
2479 https://bugzilla.gnome.org/show_bug.cgi?id=775110
2481 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
2483 * gst/rtsp-server/rtsp-stream.c:
2484 stream: block the output of rtpbin instead of the source pipeline
2485 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
2486 detection of the srtp rollover counter to add to the SDP.
2487 Unfortunately, it was incomplete for live pipelines where the logic
2488 blocks the source bin before creating the SDP and thus would never have
2489 the necessary informaiton to create a correct SDP with srtp encryption.
2490 Move the pad blocks to rtpbin's output pads instead so that the
2491 necessary information can be created before we need the information for
2493 https://bugzilla.gnome.org/show_bug.cgi?id=770239
2495 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
2497 * gst/rtsp-server/rtsp-client.c:
2498 rtsp-client: add IDLE timeout, before session exists
2499 The RTSP server will not timeout an idle RTSP connection
2500 (note this is different from doing timeout on a RTSP
2502 At least for Apache this is a problem when running RTSP over
2503 HTTPS since it uses one of the threads (there is a rather
2504 limited number) that are available for handling requests.
2505 https://bugzilla.gnome.org/show_bug.cgi?id=771830
2507 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
2512 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
2514 * gst/rtsp-server/rtsp-stream.c:
2515 rtsp-stream: Set close-socket FALSE on UDP src:es
2516 With this RTSP server can use the sockets independent on the udpsrc
2518 When the udp src is finalized it will unref socket and when g_socket
2519 is finalized the socket will be closed.
2520 https://bugzilla.gnome.org/show_bug.cgi?id=765673
2522 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2524 * gst/rtsp-sink/gstrtspclientsink.c:
2525 rtspclientsink: Move to new helper function to parse authentication responses
2526 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2528 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2530 * examples/Makefile.am:
2531 * examples/test-auth-digest.c:
2532 * gst/rtsp-server/rtsp-auth.c:
2533 * gst/rtsp-server/rtsp-auth.h:
2534 * win32/common/libgstrtspserver.def:
2535 rtsp-auth: Add support for Digest authentication
2536 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2538 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2541 * gst/rtsp-server/meson.build:
2543 * tests/check/meson.build:
2545 * win32/common/libgstrtspserver.def:
2546 Enable building with MSVC
2547 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2549 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2552 meson: gstreamer gst_check_dep does not exist on windows
2554 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2556 * gst/rtsp-server/rtsp-client.c:
2557 client: update do_send_message to match type GstRTSPClientSendFunc
2558 This type mismatch fails building with MSVC
2559 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2561 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2563 * gst/rtsp-server/rtsp-sdp.c:
2564 rtsp-sdp: Fix indentation
2566 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
2568 * gst/rtsp-server/rtsp-media.c:
2569 rtsp-media: Only signal "new-state" if the state has actually changed
2570 https://bugzilla.gnome.org/show_bug.cgi?id=774173
2572 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
2574 * gst/rtsp-server/rtsp-client.c:
2575 * gst/rtsp-server/rtsp-client.h:
2576 client: emit signal in the beginning of each rtsp request
2577 These signals let the application validate the requests, configure the
2578 media/stream in a certain way and also generate error status code in
2579 case of error or bad request.
2580 https://bugzilla.gnome.org/show_bug.cgi?id=758062
2582 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
2585 meson: update version
2587 === release 1.11.0 ===
2589 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2594 === release 1.10.0 ===
2596 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2602 * gst-rtsp-server.doap:
2605 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2607 * tests/check/gst/rtspserver.c:
2608 * tests/check/gst/stream.c:
2609 tests: try to avoid using the same ports in different tests
2610 Causes problems with client multicast tests otherwise if
2611 tests are run in parallel.
2612 https://bugzilla.gnome.org/show_bug.cgi?id=773640
2614 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2616 * tests/check/gst/client.c:
2617 tests: client: use fail_unless_equals_foo() for better failure reporting
2619 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
2621 * gst/rtsp-server/rtsp-client.c:
2622 rtsp-client: Session filter in unwatch session
2623 Call session filter with filter_session_media as paramer in
2624 client_unwatch_session if using drop_backlog = FALSE.
2625 In client_unwatch_session its allowed to grow the watchs backlog.
2626 If using drop_backlog = FALSE and the backlog is full it will cause
2627 a deadlock when setting session media state to NULL
2628 if the backlog is not allowed to grow.
2629 https://bugzilla.gnome.org/show_bug.cgi?id=771983
2631 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2634 meson: add fallbacks for gst modules
2637 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
2639 * gst/rtsp-server/rtsp-client.c:
2640 rtsp-client: Fix factory leaking in find_media() in error cases
2641 https://bugzilla.gnome.org/show_bug.cgi?id=771488
2643 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2645 * gst/rtsp-server/rtsp-stream.c:
2646 stream: Fix randomly missing streams from SDP with dynamic elements
2647 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
2648 "pad-added" signal. In that case priv->srcpad could already have its caps,
2649 and they'll be sent to priv->send_src[0] pad. That means that when it
2650 connects "notify::caps" signal, that pad could already have received its
2651 caps and the signal won't be emitted anymore.
2652 In that case priv->caps stay to NULL and when building the SDP that stream
2653 gets ignored. Leading to missing video or audio when playing in client side.
2654 https://bugzilla.gnome.org/show_bug.cgi?id=772478
2656 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
2659 meson: update version
2661 === release 1.9.90 ===
2663 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2669 * gst-rtsp-server.doap:
2672 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
2674 * gst/rtsp-server/rtsp-media-factory.c:
2675 * gst/rtsp-server/rtsp-media.c:
2676 * gst/rtsp-server/rtsp-stream.c:
2677 rtsp-server: Hint that set_multicast_iface expects the name of the interface
2678 To prevent any possibly confusion with IPs or anything else.
2679 https://bugzilla.gnome.org/show_bug.cgi?id=771530
2681 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
2683 * gst/rtsp-server/rtsp-media-factory.c:
2684 * gst/rtsp-server/rtsp-media.c:
2685 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
2686 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2688 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2691 configure: Depend on gstreamer 1.9.2.1
2693 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
2697 Automatic update of common submodule
2698 From b18d820 to f980fd9
2700 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
2704 Automatic update of common submodule
2705 From 6f2d209 to b18d820
2707 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2709 * gst/rtsp-server/rtsp-stream.c:
2710 rtsp-stream: Remove unused _locked() variant of a function
2711 It was added during refactoring.
2713 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2715 * gst/rtsp-server/rtsp-stream.c:
2716 stream: cosmetic cleanup
2717 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2719 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2721 * gst/rtsp-server/rtsp-stream.c:
2722 stream: Compare IP addresses case insensitive in more places
2723 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2725 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2728 * gst/rtsp-server/rtsp-stream.c:
2729 stream: Fix leaked joined_bin
2730 There is no need to keep a strong ref on it, and _leave_bin() was
2731 setting it to NULL before calling g_clear_object() so it was leaked.
2732 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2734 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2736 * gst/rtsp-server/rtsp-stream.c:
2737 rtsp-stream: Compare IP address strings case insensitive
2738 Otherwise IPv6 addresses might fail this comparision.
2740 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
2742 * gst/rtsp-server/rtsp-stream.c:
2743 rtsp-stream: Bind multicast sockets to ANY as before
2744 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2746 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
2748 * gst/rtsp-server/rtsp-session.c:
2749 rtsp-session: Fix segfault when doing keep-alive after removing the session
2750 If keep-alive happens after removing the session but before finalizing the
2751 stream transport, we would segfault.
2752 https://bugzilla.gnome.org/show_bug.cgi?id=750544
2754 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
2756 * gst/rtsp-server/rtsp-stream.c:
2757 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
2758 Adding them later will cause deadlocks due to
2759 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2760 2) adding the multicast sink
2761 3) waiting for it to get data to preroll again
2762 3) never happens because the queues after the tee are full.
2764 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
2766 * gst/rtsp-server/rtsp-stream.c:
2767 rtsp-stream: Fix up various multicast related issues
2769 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
2771 * tests/check/gst/stream.c:
2772 tests: Fix compilation
2774 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2776 * gst/rtsp-server/rtsp-client.c:
2777 * gst/rtsp-server/rtsp-stream.c:
2778 * tests/check/gst/stream.c:
2779 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
2780 This is basically reverting changes introduced in commit f62a9a7,
2781 because it was introducing various regressions:
2782 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
2783 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
2784 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
2785 - If a mcast client connects, it creates a new socket in SETUP to try to respect
2786 the destination/port given by the client in the transport, and overrides the
2787 socket already set on the udpsink element. That means that if we already had a
2788 client connected, the source address on the udp packets it receives suddenly
2790 - If a 2nd mcast client connects, the destination/port in its transport is
2791 ignored but its transport wasn't updated.
2792 What this patch does:
2793 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
2794 - Always have a tee+queue when udp is enabled. This could be optimized
2795 again in a later patch, but is more complicated. If no unicast clients
2796 connects then those elements are useless, this could be also optimized
2798 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
2799 seperated from those for unicast clients. Since we already support only
2800 one mcast address, we also create only one set of elements.
2801 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2803 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2805 * gst/rtsp-server/rtsp-stream.c:
2806 stream: factor our plug_src function
2807 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2809 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2811 * gst/rtsp-server/rtsp-stream.c:
2812 stream: factor out plug_sink function
2813 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2815 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2817 * gst/rtsp-server/rtsp-stream.c:
2818 stream: small documentation clarification
2819 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2821 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2823 * gst/rtsp-server/rtsp-stream.c:
2824 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
2825 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2827 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2829 * gst/rtsp-server/rtsp-stream.c:
2830 stream: Keep a ref on joined bin
2831 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2833 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2835 * gst/rtsp-server/rtsp-stream.c:
2836 stream: code cleanup
2837 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2839 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2841 * gst/rtsp-server/rtsp-stream.c:
2842 stream: small fix in error code path
2843 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2845 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2847 * gst/rtsp-server/rtsp-stream.c:
2848 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
2849 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
2850 but keeps unit tests.
2851 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2853 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
2858 === release 1.9.2 ===
2860 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2866 * gst-rtsp-server.doap:
2869 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
2872 * examples/meson.build:
2874 * gst/rtsp-server/meson.build:
2875 * gst/rtsp-sink/meson.build:
2877 * pkgconfig/meson.build:
2878 * tests/check/meson.build:
2879 * tests/meson.build:
2880 Add support for Meson as alternative/parallel build system
2881 https://github.com/mesonbuild/meson
2883 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
2886 * tests/check/Makefile.am:
2887 build: silence error about pthread for 'make check' in osx
2888 Fixes "clang: error: argument unused during compilation: '-pthread'"
2890 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
2892 * gst/rtsp-server/rtsp-client.c:
2893 rtsp-client: Fix leaking of media in error cases
2894 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
2895 and myself to make the media refcounting a bit easier to follow.
2896 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2898 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2900 * gst/rtsp-server/rtsp-client.c:
2901 rtsp-client: Fix leaking of session in error cases
2902 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2904 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
2907 Automatic update of common submodule
2908 From f363b32 to f49c55e
2910 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
2915 === release 1.9.1 ===
2917 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2923 * gst-rtsp-server.doap:
2926 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2929 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
2930 https://bugzilla.gnome.org/show_bug.cgi?id=767463
2932 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2935 Automatic update of common submodule
2936 From ac2f647 to f363b32
2938 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2940 * gst/rtsp-server/rtsp-sdp.c:
2941 * gst/rtsp-server/rtsp-sdp.h:
2942 * gst/rtsp-server/rtsp-stream.c:
2943 * gst/rtsp-server/rtsp-stream.h:
2944 sdp: add rollover counters for all sender SSRC
2945 We add different crypto sessions in MIKEY, one for each sender
2946 SSRC. Currently, all of them will have the same security policy, 0.
2947 The rollover counters are obtained from the srtpenc element using the
2949 https://bugzilla.gnome.org/show_bug.cgi?id=730539
2951 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2953 * gst/rtsp-server/rtsp-media-factory.h:
2954 * gst/rtsp-server/rtsp-server.h:
2955 docs: fix some typos
2957 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
2959 * gst/rtsp-server/Makefile.am:
2960 g-i: pass compiler env to g-ir-scanner
2961 It's what introspection.mak does as well. Should
2962 fix spurious build failures on gnome-continuous
2963 (caused by g-ir-scanner getting compiler details
2964 via python which is broken in some environments
2965 so passing the compiler details bypasses that).
2967 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
2969 * gst/rtsp-server/rtsp-session.c:
2970 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
2971 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
2972 https://bugzilla.gnome.org/show_bug.cgi?id=766619
2974 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
2976 * gst/rtsp-sink/gstrtspclientsink.c:
2977 rtspclientsink: Check return value of sscanf
2978 And just make sure we always have 0/0 if we have an error
2981 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
2983 * gst/rtsp-server/rtsp-stream.c:
2984 * tests/check/gst/rtspserver.c:
2985 * tests/check/gst/stream.c:
2986 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
2987 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
2988 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
2989 - Create unit test for shared media.
2990 https://bugzilla.gnome.org/show_bug.cgi?id=764744
2992 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2994 * gst/rtsp-server/rtsp-stream.c:
2995 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
2996 For IPv6 addresses, binding to a multicast group does not work on Linux
2997 either. Always bind to ANY and then later join the multicast group.
2998 https://bugzilla.gnome.org/show_bug.cgi?id=764679
3000 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
3003 Automatic update of common submodule
3004 From 6f2d209 to ac2f647
3006 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
3008 * gst/rtsp-server/rtsp-thread-pool.c:
3009 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
3010 Clarified why it is necessary to add source information to
3011 GstRTSPThreadImpl. See the reported bug in GLib:
3012 https://bugzilla.gnome.org/show_bug.cgi?id=720186
3013 for more information.
3014 https://bugzilla.gnome.org/show_bug.cgi?id=761702
3016 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
3018 * examples/Makefile.am:
3019 examples: Clean up CFLAGS/LDADD even more
3020 The internal .la should come first and is part of LDADD, as is
3023 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
3025 * examples/Makefile.am:
3026 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
3028 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
3030 * gst/rtsp-server/Makefile.am:
3031 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
3033 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3035 * gst/rtsp-server/rtsp-client.c:
3036 * gst/rtsp-server/rtsp-media-factory.c:
3037 * gst/rtsp-server/rtsp-media-factory.h:
3038 * gst/rtsp-server/rtsp-media.c:
3039 * gst/rtsp-server/rtsp-media.h:
3040 * gst/rtsp-server/rtsp-sdp.c:
3041 * gst/rtsp-server/rtsp-stream.c:
3042 * gst/rtsp-server/rtsp-stream.h:
3043 rtsp-server: Implement clock signalling according to RFC7273
3044 For NTP and PTP clocks we signal the actual clock that is used and signal
3045 the direct media clock offset.
3046 For all other clocks we at least signal that it's the local sender clock.
3047 This allows receivers to know which clock was used to generate the media and
3048 its RTP timestamps. Receivers can then implement network synchronization,
3049 either absolute or at least relative by getting the sender clock rate directly
3050 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
3052 https://bugzilla.gnome.org/show_bug.cgi?id=760005
3054 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
3056 * gst/rtsp-sink/gstrtspclientsink.c:
3057 rtspclientsink: Add support for setting the multicast interface
3058 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3060 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3062 * gst/rtsp-server/rtsp-media-factory.c:
3063 * gst/rtsp-server/rtsp-media-factory.h:
3064 * gst/rtsp-server/rtsp-media.c:
3065 * gst/rtsp-server/rtsp-media.h:
3066 * gst/rtsp-server/rtsp-stream.c:
3067 * gst/rtsp-server/rtsp-stream.h:
3068 rtsp-media: Add support for setting the multicast interface
3069 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3071 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
3073 * gst/rtsp-sink/gstrtspclientsink.c:
3074 rtspclientsink: use new gst_element_class_add_static_pad_template()
3075 https://bugzilla.gnome.org/show_bug.cgi?id=763196
3077 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3082 === release 1.8.0 ===
3084 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
3090 * gst-rtsp-server.doap:
3093 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
3095 * gst/rtsp-server/rtsp-stream.c:
3096 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
3097 This would get us NO_PREROLL in the bin again and break seeking.
3098 Thanks to Carlos Rafael Giani for helping to debug this!
3099 https://bugzilla.gnome.org/show_bug.cgi?id=740509
3101 === release 1.7.91 ===
3103 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3109 * gst-rtsp-server.doap:
3112 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3114 * gst/rtsp-server/rtsp-stream.c:
3115 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
3116 Without this, RECORD pipelines are broken because
3117 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
3118 added later. Previously it was there earlier and due to NO_PREROLL caused the
3119 pipeline to preroll immediately
3120 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
3121 as the corresponding code previously was only for PLAY pipelines.
3122 https://bugzilla.gnome.org/show_bug.cgi?id=763281
3124 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
3126 * gst/rtsp-server/rtsp-stream.c:
3127 rtsp-stream: Fix typo in the docstring
3128 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
3130 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
3132 * gst/rtsp-server/rtsp-stream.c:
3133 rtsp-stream: Disable multicast loopback for all our sockets
3134 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
3135 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
3136 loopback setting on the socket... while udpsink does which unfortunately has
3137 no effect here on Windows but on Linux.
3138 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3140 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
3142 * tests/check/gst/stream.c:
3143 stream tests: added new tests
3144 Test a case when the address pool only contains multicast addresses
3145 and the client is requesting unicast udp.
3146 Added tests for multicast ports allocation.
3147 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3149 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
3151 * gst/rtsp-server/rtsp-stream.c:
3152 rtsp-stream: Only bind multicast sockets to ANY on Windows
3153 On Linux it is still needed to bind to the multicast address
3154 to filter out random other packets, while on Windows binding
3155 to multicast addresses just fails.
3157 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3159 * gst/rtsp-server/rtsp-stream.c:
3160 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
3161 Otherwise we fail to allocate UDP ports if the pool only contains multicast
3162 addresses, which is something that used to work before. For unicast addresses
3163 if the pool contains none, we just allocate them as if there is no pool at
3165 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3167 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
3169 * gst/rtsp-server/rtsp-client.c:
3170 * gst/rtsp-server/rtsp-stream.c:
3171 rtsp-server: Fix indentation
3173 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
3175 * gst/rtsp-server/rtsp-stream.c:
3176 rtsp-stream: Don't bind the sockets to multicast addresses
3177 This works on Linux but fails completely on Windows. You're supposed
3178 to bind to ANY and then join the multicast group.
3179 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3181 === release 1.7.90 ===
3183 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3189 * gst-rtsp-server.doap:
3192 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3195 Automatic update of common submodule
3196 From b64f03f to 6f2d209
3198 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
3200 * gst/rtsp-sink/gstrtspclientsink.c:
3201 * tests/check/gst/rtspclientsink.c:
3202 rtspsink: Fix some leaks in rtspclientsink and the unit test.
3203 https://bugzilla.gnome.org/show_bug.cgi?id=762525
3205 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
3207 * tests/check/gst/media.c:
3208 * tests/check/gst/rtspclientsink.c:
3209 * tests/check/gst/rtspserver.c:
3210 * tests/check/gst/stream.c:
3211 tests: unit test fixes
3212 Removed port allocation test from the media suite.
3213 The port allocation failure is now in the stream suite.
3215 Make sure that the media is suspended after the DESCRIBE request
3216 before reconfiguring the UDP sinks.
3218 In the RECORD case we have to set async property to false
3219 for the appsink element in the test in order to make sure
3220 that the media pipeline doesn't hang in start_preroll().
3221 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3223 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
3225 * gst/rtsp-server/rtsp-client.c:
3226 * gst/rtsp-server/rtsp-stream.c:
3227 * gst/rtsp-server/rtsp-stream.h:
3228 rtsp-stream: postpone UDP socket allocation until SETUP
3229 Postpone the allocation of the UDP sockets until we know
3230 what transport has been chosen by the client.
3231 Both unicast and multicast UDP sources are created in one
3233 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3235 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
3237 * gst/rtsp-server/rtsp-stream.c:
3238 rtsp-stream: postpone the creation of the UDP sources
3239 Code refactoring: allocate the UDP ports after the sender and
3240 the reciver parts have been created.
3241 We postpone the creation of the UDP sources until the UDP
3242 ports have been allocated.
3243 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3245 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
3247 * gst/rtsp-server/rtsp-stream.c:
3248 rtsp-stream: added function for setting UDP sources to PLAYING state
3249 Code refactoring: Introduced a function for setting UDP sources
3251 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3253 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
3255 * gst/rtsp-server/rtsp-stream.c:
3256 rtsp-stream: added function for creating and configuring UDP sources
3257 Code refactoring: create and configure UDP sources in a separate function.
3258 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3260 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
3262 * gst/rtsp-server/rtsp-stream.c:
3263 rtsp-stream: added function for RTP/RTCP socket configuration
3264 Code refactoring: configure RTP and RTCP sockets for UDP sinks
3265 in a separate function.
3266 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3268 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
3270 * gst/rtsp-server/rtsp-stream.c:
3271 rtsp-stream: added function for creating and configuring UDP sinks
3272 Code refactoring: create and configure UDP sinks in a separate function.
3273 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3275 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
3277 * gst/rtsp-server/rtsp-stream.c:
3278 rtsp-stream: added helper function for creating the sender/receiver parts
3279 Code refactoring: introduced helper function for creating
3280 the receiver and the sender parts of the streaming pipeline.
3281 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3283 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
3288 === release 1.7.2 ===
3290 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
3296 * gst-rtsp-server.doap:
3299 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
3301 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3302 uninstalled.pc: add support for non libtool build systems
3303 Currently the .la path is provided which requires to use libtool as
3304 mentioned in the GStreamer manual section-helloworld-compilerun.html.
3305 It is fine as long as the application is built using libtool.
3306 So currently it is not possible to compile a GStreamer application
3307 within gst-uninstalled with CMake or other build system different
3309 This patch allows to do the following in gst-uninstalled env:
3310 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
3311 gstreamer-rtsp-server-1.0)
3312 Previously it required to prepend libtool --mode=link
3313 https://bugzilla.gnome.org/show_bug.cgi?id=720778
3315 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3317 * gst/rtsp-sink/gstrtspclientsink.c:
3318 rtspclientsink: remove check for impossible condition
3319 Goto error label checks stream to see if it needs to be unreferenced before
3320 returning, but this goto jumps happens before the stream is ever set, so it
3321 will always be NULL in this error label.
3324 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3326 * gst/rtsp-sink/gstrtspclientsink.c:
3327 rtspclientsink: clean switch statements
3328 Coverity demands for fallthrough statements to be clearly commented,
3329 to distinguish from accidental fall throughs. And it also needs all
3330 cases to finish with a break, even if the break is never going to be
3331 executed like in the case of a continue jump.
3335 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3337 * tests/check/Makefile.am:
3338 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
3339 To get the CK_DEFAULT_TIMEOUT defined for all tests
3340 Also removes a 120 seconds timeout that was set as default
3341 explicitly in this module
3342 https://bugzilla.gnome.org/show_bug.cgi?id=761472
3344 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3348 Automatic update of common submodule
3349 From 86e4663 to b64f03f
3351 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
3353 * gst/rtsp-server/rtsp-media.c:
3354 rtsp-media: fix state_lock not locked again when preroll fails
3355 https://bugzilla.gnome.org/show_bug.cgi?id=761399
3357 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
3360 configure: Move plugin specific flags below all the others
3361 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
3362 -no-undefined. And -no-undefined is required on Windows to build DLLs.
3364 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
3366 * gst/rtsp-sink/gstrtspclientsink.c:
3367 rtspclientsink: Simplify slightly using new -base API
3368 Use the new Mikey and SDP API in the base plugins libs
3369 to simplify some code.
3370 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3372 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3377 * gst/rtsp-sink/Makefile.am:
3378 * gst/rtsp-sink/gstrtspclientsink.c:
3379 * gst/rtsp-sink/gstrtspclientsink.h:
3380 * gst/rtsp-sink/plugin.c:
3381 * tests/check/Makefile.am:
3382 * tests/check/gst/rtspclientsink.c:
3383 rtspsink: Add rtspclientsink element
3384 Add an rtspclientsink element that accepts streams for which
3385 there is a registered payloader and sends them to
3386 an RTSP server using RECORD.
3387 Sending is synchronised to the pipeline clock. Payload-types
3388 are automatically selected. The 'new-payloader' signal is fired
3389 for custom configuration of payloaders when they are created.
3390 Can now stream a movie like this:
3392 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
3393 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
3395 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
3396 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
3397 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3399 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3401 * gst/rtsp-server/rtsp-stream.c:
3402 * gst/rtsp-server/rtsp-stream.h:
3403 rtsp-stream: Add functions for using rtsp-stream from the client
3404 Add a boolean to indicate that the rtsp-stream is running on the
3405 'client' side of an RTSP connection, for sending streams via
3406 RECORD. In that case, the roles of the client/server ports
3407 in transport setup are swapped.
3408 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3410 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3412 * gst/rtsp-server/rtsp-sdp.c:
3413 * gst/rtsp-server/rtsp-sdp.h:
3414 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
3415 A new function that adds info from a GstRTSPStream into an SDP message.
3416 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3418 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
3420 * gst/rtsp-server/rtsp-media.c:
3421 rtsp-media: Fix mutex beeing unlocked while they should be locked
3422 https://bugzilla.gnome.org/show_bug.cgi?id=761226
3424 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
3426 * gst/rtsp-server/rtsp-media-factory.c:
3427 rtsp-media-factory: add missing break in "clock" property setter
3430 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
3432 * gst/rtsp-server/rtsp-stream.c:
3433 rtsp-stream: fixed assert during update transport
3434 When RTSP server trying update transport during multicast, it throws an
3435 assert. The assert is thrown because it is trying to get the parent of
3436 an non-existing funnel element.
3437 https://bugzilla.gnome.org/show_bug.cgi?id=760150
3439 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
3441 * gst/rtsp-server/rtsp-permissions.h:
3442 * gst/rtsp-server/rtsp-thread-pool.h:
3443 * gst/rtsp-server/rtsp-token.h:
3444 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
3445 gtk-doc can handle static inline functions just fine these days,
3446 there's no need for this stuff any more.
3448 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3450 * gst/rtsp-server/rtsp-media.c:
3451 * gst/rtsp-server/rtsp-sdp.c:
3452 sdp: replace duplicated codes to call new base sdp apis
3453 https://bugzilla.gnome.org/show_bug.cgi?id=745880
3455 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
3457 * examples/test-netclock.c:
3458 test-netclock: Use the new API to configure a clock directly
3460 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3462 * gst/rtsp-server/rtsp-media-factory.c:
3463 * gst/rtsp-server/rtsp-media-factory.h:
3464 * gst/rtsp-server/rtsp-media.c:
3465 * gst/rtsp-server/rtsp-media.h:
3466 rtsp-media: Add API to directly configure a clock on the media pipelines
3468 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3470 * gst/rtsp-server/rtsp-media.c:
3471 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
3473 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3475 * gst/rtsp-server/rtsp-media-factory.c:
3476 rtsp-media-factory: Add FIXME for 2.0
3478 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3480 * gst/rtsp-server/rtsp-stream.c:
3481 rtsp-stream: Fix indentation
3483 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3485 * gst/rtsp-server/rtsp-media.c:
3486 rtsp-media: Do not prepare media after media times out
3487 Deferred calls to start_prepare() can be deferred past the point until
3488 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
3489 prepared to wait. Previously there was no lock and no check for this
3490 situation. This meant that a media could be prepared and unprepared
3491 simultaneously by two different threads. Now a lock is in place and a
3492 suitable check is done.
3493 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
3495 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3497 * gst/rtsp-server/rtsp-client.c:
3498 * gst/rtsp-server/rtsp-media-factory.c:
3499 * gst/rtsp-server/rtsp-media-factory.h:
3500 * gst/rtsp-server/rtsp-media.c:
3501 * gst/rtsp-server/rtsp-media.h:
3502 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
3503 Without TEARDOWN it might be desireable to keep the media running and continue
3504 sending data to the client, even if the RTSP connection itself is
3506 Only do this for session medias that have only UDP transports. If there's at
3507 least on TCP transport, it will stop working and cause problems when the
3508 connection is disconnected.
3509 https://bugzilla.gnome.org/show_bug.cgi?id=758999
3511 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
3516 === release 1.7.1 ===
3518 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3524 * gst-rtsp-server.doap:
3527 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
3530 configure: Make -Bsymbolic check work with clang.
3531 Update the -Bsymbolic check with the version glib has. This version
3533 https://bugzilla.gnome.org/show_bug.cgi?id=759713
3535 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3537 * gst/rtsp-server/rtsp-session-pool.c:
3538 rtsp-session-pool: Avoid dollar sign ($) in session ids
3539 Live555 in VLC strips off dollar signs and then gets very confused,
3540 we don't loose too much entropy by just skipping it.
3542 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
3544 * gst/rtsp-server/rtsp-address-pool.h:
3545 * gst/rtsp-server/rtsp-auth.h:
3546 * gst/rtsp-server/rtsp-client.h:
3547 * gst/rtsp-server/rtsp-media-factory-uri.h:
3548 * gst/rtsp-server/rtsp-media-factory.h:
3549 * gst/rtsp-server/rtsp-media.h:
3550 * gst/rtsp-server/rtsp-mount-points.h:
3551 * gst/rtsp-server/rtsp-permissions.h:
3552 * gst/rtsp-server/rtsp-server.h:
3553 * gst/rtsp-server/rtsp-session-media.h:
3554 * gst/rtsp-server/rtsp-session-pool.h:
3555 * gst/rtsp-server/rtsp-session.h:
3556 * gst/rtsp-server/rtsp-stream-transport.h:
3557 * gst/rtsp-server/rtsp-stream.h:
3558 * gst/rtsp-server/rtsp-thread-pool.h:
3559 * gst/rtsp-server/rtsp-token.h:
3560 rtsp-server: Add g_autoptr() support to all types
3561 https://bugzilla.gnome.org/show_bug.cgi?id=754464
3563 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
3565 * gst/rtsp-server/rtsp-stream.c:
3566 rtsp-stream: fixed valgrind error
3567 Fixed the valgrind error in unit test. The UDP source created during
3568 gst_rtsp_stream_join_bin() was not released while destroying the rtp
3570 https://bugzilla.gnome.org/show_bug.cgi?id=759010
3572 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3576 Automatic update of common submodule
3577 From b319909 to 86e4663
3579 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
3581 * gst/rtsp-server/rtsp-client.c:
3582 rtsp-client: suspend media during setup request
3583 SETUP request from clients needs to suspend the media to clear the
3584 prerolled buffers. Otherwise it will not affect the prerolled buffer
3585 and the prerolled buffers will be incorrect (for example block-size
3586 from setup request will not affect the prerolled buffer unless the
3587 media is suspended).
3588 https://bugzilla.gnome.org/show_bug.cgi?id=758268
3590 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
3592 * gst/rtsp-server/rtsp-stream.c:
3593 rtsp-stream: create stream pipeline based on transport
3594 Based on the protocol, create the rtsp stream pipeline. If only TCP or
3595 only UDP is set as the transport protocol, it will not add the extra tee
3596 or queue element to the pipeline. Both these elements will be added, if
3597 it supports both TCP and UDP protocols. This improves the pipeline
3598 performance when one protocol is present.
3599 https://bugzilla.gnome.org/show_bug.cgi?id=758179
3601 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3603 * gst/rtsp-server/rtsp-stream.c:
3604 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
3605 Adding them when not needed will start some logic inside rtpbin that might be
3606 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
3607 would start up a rtpjitterbuffer and behave in weird ways.
3608 We still set up the UDP sources for RTP receiving for a sender media to be
3609 able to receive any packets sent by the client for NAT traversal. They will
3610 all go to a fakesink though.
3611 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
3612 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
3613 receive ASYNC_DONE after a seek.
3614 https://bugzilla.gnome.org/show_bug.cgi?id=758319
3616 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3618 * gst/rtsp-server/rtsp-stream.c:
3619 rtsp-stream: Disable multicast loopback for the multicast udp sources too
3620 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
3621 Previously we were only setting this for sender sockets, which caused looped
3622 back packets to be received on Windows if a multicast transport was used.
3624 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3626 * examples/test-record-auth.c:
3627 * examples/test-record.c:
3628 examples: Actually use the provided port in the record examples
3630 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3632 * examples/test-record-auth.c:
3633 test-record-auth: Add the option to build in TLS support
3635 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3637 * examples/test-auth.c:
3638 test-auth: Use an 'anonymous' user for unauthenticated default
3639 There's a comment on one of the resources that 'user' and 'admin'
3640 shouldn't even be able to see it, but they can if the default
3641 token is 'admin2', since that gives them access anyway.
3643 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3645 * examples/.gitignore:
3646 * examples/Makefile.am:
3647 * examples/test-record-auth.c:
3648 Add test-record-auth example
3650 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3652 * gst/rtsp-server/rtsp-client.c:
3653 * tests/check/gst/client.c:
3654 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
3656 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
3658 * gst/rtsp-server/rtsp-server.c:
3659 rtsp-server: Change the logic so we don't pop a NULL context
3660 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
3661 will sometimes fail. This call is made before any context is pushed
3662 resulting in an attempt to pop a NULL context.
3663 https://bugzilla.gnome.org/show_bug.cgi?id=757949
3665 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
3667 * tests/check/gst/rtspserver.c:
3668 rtspserver: Add udp-mcast transport SETUP test
3669 Refactor utility functions in the test file so they can handle
3670 more than UDP and TCP as lower transport.
3671 https://bugzilla.gnome.org/show_bug.cgi?id=756969
3673 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
3675 * gst/rtsp-server/rtsp-stream.c:
3676 rtsp-stream: Always unref return value of gst_object_get_parent()
3677 Fixes a leak of a GstBin in the udp-mcast case.
3678 https://bugzilla.gnome.org/show_bug.cgi?id=756968
3680 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
3683 Automatic update of common submodule
3684 From b99800a to b319909
3686 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
3689 Use new GST_ENABLE_EXTRA_CHECKS #define
3690 https://bugzilla.gnome.org/show_bug.cgi?id=756870
3692 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3695 Automatic update of common submodule
3696 From 6babecd to b99800a
3698 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3701 Update GLib dependency to 2.40.0
3703 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3705 * examples/test-mp4.c:
3706 * gst/rtsp-server/rtsp-stream.c:
3707 stream: listen to sender ssrc signals
3708 https://bugzilla.gnome.org/show_bug.cgi?id=746747
3710 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
3713 common: update for new suppression
3714 Makes check-valgrind pass with glib 2.46
3716 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3718 * gst/rtsp-server/rtsp-media.c:
3719 rtsp-media: Take reference to media that will be prepared
3720 default_prepare() takes a transfer-none reference GstRTSPMedia object.
3721 Later on a g_idle_source_new() is created and a pointer to the media
3722 object is passed as user data. If the media is freed before the idle
3723 source is dispatched the media object pointer is invalid, but the idle
3724 source callback expects it to still be valid. To fix this a reference to
3725 the media object is taken when registering the source callback function
3726 and a corresponding release of the reference is done when the souce is
3728 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
3730 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
3732 * examples/test-launch.c:
3733 * examples/test-mp4.c:
3734 * examples/test-ogg.c:
3735 * examples/test-record.c:
3736 * examples/test-uri.c:
3737 rtsp-server: Fix memory leaks when context parse fails
3738 When g_option_context_parse fails, context and error variables are not getting free'd
3739 which results in memory leaks. Free'ing the same.
3740 And replacing g_error_free with g_clear_error, which checks if the error being passed
3741 is not NULL and sets the variable to NULL on free'ing.
3742 https://bugzilla.gnome.org/show_bug.cgi?id=753863
3744 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3749 === release 1.6.0 ===
3751 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3757 * gst-rtsp-server.doap:
3760 === release 1.5.91 ===
3762 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
3768 * gst-rtsp-server.doap:
3771 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
3773 * docs/libs/gst-rtsp-server-sections.txt:
3774 * gst/rtsp-server/rtsp-stream.c:
3775 stream: fix docs for recently-added get/set_buffer_size API
3776 https://bugzilla.gnome.org/show_bug.cgi?id=749095
3778 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
3780 * gst/rtsp-server/rtsp-media.c:
3781 rtsp-media: Don't crash on encrypted RTX SDP
3782 In parse_keymgmt(), don't mutate the input string that's been passed
3783 as const, especially since we might need the original value again if
3784 the same key info applies to multiple streams (RTX, for example).
3785 https://bugzilla.gnome.org/show_bug.cgi?id=754753
3787 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
3789 * examples/test-mp4.c:
3790 test-mp4: Support filenames with spaces in them. Error out on too few arguments
3792 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
3794 * examples/test-record.c:
3795 test-record: Check parameter count and print out help
3796 If no launch pipeline was supplied, print out some help
3798 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
3800 * gst/rtsp-server/rtsp-media.c:
3801 * gst/rtsp-server/rtsp-stream.c:
3802 * gst/rtsp-server/rtsp-stream.h:
3803 rtsp-stream: Implement UDP buffer size setting.
3804 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
3806 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
3807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
3809 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
3811 * gst/rtsp-server/rtsp-media.h:
3812 rtsp-media: Fix small typo causing gtk-doc to complain
3814 === release 1.5.90 ===
3816 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3822 * gst-rtsp-server.doap:
3825 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3827 * gst/rtsp-server/rtsp-media-factory.c:
3828 media-factory: get port number through gst_rtsp_url_get_port
3829 https://bugzilla.gnome.org/show_bug.cgi?id=753473
3831 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
3833 * tests/check/gst/media.c:
3834 media-test: Removing unnecessary assertion
3835 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3837 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3839 * gst/rtsp-server/rtsp-server.c:
3840 Document that source keeps a ref on server until it's destroyed
3841 https://bugzilla.gnome.org/show_bug.cgi?id=749227
3843 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3845 * tests/check/gst/media.c:
3846 media-test: Test for multiple dynamic payload
3847 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3849 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3851 * gst/rtsp-server/rtsp-media.c:
3852 media: Only add fakesink once per pipeline
3853 The intention is to prevent going PLAYING state before pads are created.
3854 If there was mutilple dynamic payload, it would leak few fakesink and
3855 actually prevent from ever reaching playing state.
3856 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3858 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3860 * gst/rtsp-server/rtsp-media.c:
3861 Revert "rtsp-media: Only add 1 fakesink per pipeline"
3862 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
3864 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3866 * gst/rtsp-server/rtsp-media.c:
3867 rtsp-media: Only add 1 fakesink per pipeline
3868 There should be only one fakesink per pipeline, not per dynpay. This
3869 would lead to element naming clash.
3871 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
3873 * gst/rtsp-server/rtsp-media.c:
3874 rtsp-media: assertion error due to wrong condition check
3875 In media to caps function, reserved_keys array is being used for variable i,
3876 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
3877 changed it to variable j
3878 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3880 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
3882 * gst/rtsp-server/rtsp-media.c:
3883 rtsp-media: Strip keys from the fmtp that we use internally in our caps
3884 Skip keys from the fmtp, which we already use ourselves for the
3885 caps. Some software is adding random things like clock-rate into
3886 the fmtp, and we would otherwise here set a string-typed clock-rate
3887 in the caps... and thus fail to create valid RTP caps
3888 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3890 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3892 * gst/rtsp-server/rtsp-thread-pool.c:
3893 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
3894 https://bugzilla.gnome.org/show_bug.cgi?id=752640
3896 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
3899 Automatic update of common submodule
3900 From f74b2df to 9aed1d7
3902 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
3907 === release 1.5.2 ===
3909 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3915 * gst-rtsp-server.doap:
3918 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
3920 * gst/rtsp-server/rtsp-client.c:
3921 * gst/rtsp-server/rtsp-client.h:
3922 * tests/check/gst/client.c:
3923 rtsp-client: allow application to decide what requirements are supported
3924 Add "check-requirements" signal and vfunc to allow application
3925 (and subclasses) to check the requirements.
3926 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
3927 https://bugzilla.gnome.org/show_bug.cgi?id=749417
3929 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3932 Automatic update of common submodule
3933 From 6015d26 to f74b2df
3935 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
3937 * gst/rtsp-server/rtsp-media.c:
3938 rtsp-media: Always use real payloader when creating streams
3939 A bin that contains the real payloader might be used as payloader. In this
3940 case we have to get the real payloader for the various properties it provides.
3941 Example use cases for this are bins that payload some media and then have
3942 additional elements that add metadata or RTP extension headers to the stream.
3943 https://bugzilla.gnome.org/show_bug.cgi?id=750800
3945 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3947 * examples/test-netclock-client.c:
3948 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
3950 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
3952 * examples/test-netclock-client.c:
3953 * examples/test-netclock.c:
3954 test-netclock: Use new ntp-time-source property on rtpbin
3955 Select the clock time to be used as NTP time source. This allows proper
3956 synchronization between receivers, independent of sharing base times, and just
3957 requires them to use the same clock.
3959 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3961 * examples/test-netclock-client.c:
3962 * examples/test-netclock.c:
3963 test-netclock: Setting the same base time on sender and receiver is not necessary
3964 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
3966 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3968 * gst/rtsp-server/rtsp-stream.c:
3969 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
3970 https://bugzilla.gnome.org/show_bug.cgi?id=750764
3972 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3974 * docs/libs/gst-rtsp-server.types:
3975 docs: add missing types
3976 https://bugzilla.gnome.org/show_bug.cgi?id=750764
3978 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3980 * docs/libs/gst-rtsp-server-sections.txt:
3981 docs: add missing apis
3982 https://bugzilla.gnome.org/show_bug.cgi?id=750764
3984 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
3986 * examples/test-netclock-client.c:
3987 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
3989 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3991 * docs/libs/gst-rtsp-server-sections.txt:
3992 * gst/rtsp-server/rtsp-auth.c:
3993 * gst/rtsp-server/rtsp-auth.h:
3994 GstRTSPAuth: Add client certificate authentication support
3995 https://bugzilla.gnome.org/show_bug.cgi?id=750471
3997 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
3999 * examples/test-netclock-client.c:
4000 test-netclock-client: Use new GstClock API to wait for clock synchronization
4002 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
4004 * examples/test-netclock-client.c:
4005 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
4006 A mainloop is needed to get glimagesink to display something on OSX, and
4007 the source-setup signal just makes things a little bit easier.
4009 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
4012 Automatic update of common submodule
4013 From d9a3353 to 6015d26
4015 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
4018 Automatic update of common submodule
4019 From d37af32 to d9a3353
4021 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
4024 Automatic update of common submodule
4025 From 21ba2e5 to d37af32
4027 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
4030 Automatic update of common submodule
4031 From c408583 to 21ba2e5
4033 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
4035 * docs/libs/Makefile.am:
4036 docs: remove variables that we define in the snippet from common
4037 This is syncing our Makefile.am with upstream gtkdoc.
4039 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4042 Automatic update of common submodule
4043 From 44a3517 to c408583
4045 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
4050 === release 1.5.1 ===
4052 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
4058 * gst-rtsp-server.doap:
4061 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
4063 * gst/rtsp-server/rtsp-client.c:
4064 rtsp-client: No flush during Teardown.
4065 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
4066 backlog is empty it can happen that just a part of a message will be
4067 sent and rest is in backlog queue. If then flush during teardown
4068 just a part of message will be sent.This can lead to client miss
4069 teardown response since it expect to get the last part of message.
4070 The flushing during teardown was introduced to fix a deadlock that now
4071 is fixed more generally in handle_request by temporary setting backlog
4073 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
4075 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
4077 * tests/check/Makefile.am:
4078 tests: Use AM_TESTS_ENVIRONMENT
4079 Needed by the new automake test runner and the
4080 current version of the common submodule.
4082 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
4084 * gst/rtsp-server/rtsp-media.h:
4085 * gst/rtsp-server/rtsp-stream.h:
4086 rtsp-server: Use single-include rtsp header to make sure we get all definitions
4088 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
4090 * gst/rtsp-server/rtsp-media.c:
4091 rtsp-media: Mark some more functions static
4093 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4095 * gst/rtsp-server/rtsp-media.c:
4096 rtsp-media: Only unblock the media in suspend() when actually changing the state
4097 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
4099 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4101 * examples/test-video-rtx.c:
4102 examples: Use AVPF profile for the RTX example
4104 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4106 * gst/rtsp-server/rtsp-sdp.c:
4107 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
4109 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4111 * gst/rtsp-server/rtsp-stream.c:
4112 rtsp-stream: get valid clock-rate from last-sample
4113 clock-rate in last-sample's caps is integer, not unsigned.
4114 To get this value properly, variable needs to be type-casted to int.
4115 https://bugzilla.gnome.org/show_bug.cgi?id=747614
4117 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
4121 autogen.sh: only run autopoint if gettext requested in configure.ac
4122 Not just because there happens to be a po directory.
4123 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4125 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4128 Revert "configure.ac: uncomment gettext version setup"
4129 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
4130 We don't need a gettext setup here and there's no po
4131 directory either, so no reason why autopoint would be
4132 run in the first place.
4133 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
4135 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
4137 * examples/test-multicast.c:
4138 * examples/test-multicast2.c:
4139 * examples/test-sdp.c:
4140 * examples/test-video-rtx.c:
4141 * examples/test-video.c:
4142 * tests/test-cleanup.c:
4143 * tests/test-reuse.c:
4144 Fix timeout function signatures across tests and examples
4146 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
4148 * tests/check/Makefile.am:
4149 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
4150 Make sure the test environment is set up.
4151 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4153 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4156 configure: bump automake requirement to 1.14 and autoconf to 2.69
4157 This is only required for builds from git, people can still
4158 build tarballs if they only have older autotools.
4159 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4161 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4164 configure.ac: uncomment gettext version setup
4165 Fixes autogen.sh. It would run autopoint, which would complain
4166 that it could not find the gettext version in configure.ac.
4167 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4169 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4171 * examples/test-video-rtx.c:
4172 test-video-rtx: set exact payload type to PCMA payloader
4173 Setting wrong payload type causes failure to do retransmission through audio stream
4174 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4176 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4178 * gst/rtsp-server/rtsp-media.c:
4179 * gst/rtsp-server/rtsp-stream.c:
4180 * gst/rtsp-server/rtsp-stream.h:
4181 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
4182 Because of duplicated g_signal_connect for request-aux-sender signal,
4183 wrong stream pointer is passed to the signal handler.
4184 Instead of passing each stream, pass stream array and get the relevant stream.
4185 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4187 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
4191 Update autogen.sh to latest version from common
4192 Fixes build after aclocal_check etc. helpers have been removed.
4194 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
4197 Automatic update of common submodule
4198 From bc76a8b to c8fb372
4200 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4202 * gst/rtsp-server/rtsp-stream.c:
4203 rtsp-stream: Limit the queues to 1 buffer
4204 We only need them to be able to pre-roll, queueing up more data here
4205 is only going to harm latency and memory usage.
4207 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
4209 * gst/rtsp-server/rtsp-stream.c:
4210 rtsp-stream: Update comment and ASCII art to the latest code
4211 We have a queue in front of the udpsink too to prevent the pipeline from
4214 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4216 * gst/rtsp-server/rtsp-stream.c:
4217 rtsp-media: Properly return first rtptime
4218 Instead we where returning first GstBuffer timestamp. This would result
4219 in clock skew and unwanted behaviour in RTSP playback.
4220 https://bugzilla.gnome.org/show_bug.cgi?id=746479
4222 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4224 * gst/rtsp-server/rtsp-stream.c:
4225 rtsp-stream: Don't leave buffer mapped
4226 If the seq is NULL, the RTP buffer was left mapped. We should always
4229 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
4234 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4236 * gst/rtsp-server/rtsp-media-factory.c:
4237 * tests/check/gst/client.c:
4238 Fix double semicolons
4240 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
4242 * gst/rtsp-server/rtsp-stream.c:
4243 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
4244 This gives more accurate values than asking the payloader. There might be
4245 queueing happening between the payloader and the sink.
4246 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4248 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
4250 * gst/rtsp-server/rtsp-media.c:
4251 rtsp-media: Don't seek for PLAY if the position will not change
4252 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4254 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4256 * gst/rtsp-server/rtsp-media.c:
4257 rtsp-media: Don't include payload type in the caps for framesize
4258 When the sdp media attribute framesize are converted to caps
4259 the <payload> should not be included.
4260 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
4261 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
4263 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
4265 * gst/rtsp-server/rtsp-sdp.c:
4266 rtsp-sdp: add payload type to the sdp framesize attribute
4267 The sdp framesize attribute is desribed in RFC6064. It is specified
4268 for payloading of H263 and has the following form
4269 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
4270 should be added to the caps in a payloader and the <payload type> should
4271 be added by the rtsp-server.
4272 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
4274 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4276 * examples/test-uri.c:
4277 examples: test-uri: fix tainted variable
4278 Insignificant but this keeps Coverity happy.
4281 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4283 * examples/.gitignore:
4284 * examples/Makefile.am:
4285 * examples/test-netclock-client.c:
4286 * examples/test-netclock.c:
4287 examples: Add a simple example of network synch for live streams.
4288 An example server and client that works for synchronising live streams
4289 only - as it can't support pause/play.
4291 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4293 * gst/rtsp-server/rtsp-media-factory.c:
4294 * gst/rtsp-server/rtsp-media-factory.h:
4295 rtsp-media-factory: Add functions to set/get the media gtype
4296 Allow specifying the GType of a GstRtspMedia subclass to create
4297 as a simpler way to get the factory to create a custom
4298 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
4300 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
4302 * gst/rtsp-server/rtsp-media.c:
4303 rtsp-media: fix double unlock in _get_buffer_size()
4304 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
4305 because of double g_mutex_unlock () usage.
4306 https://bugzilla.gnome.org/show_bug.cgi?id=745434
4308 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
4310 * gst/rtsp-server/rtsp-session-pool.c:
4311 * gst/rtsp-server/rtsp-session.c:
4312 * gst/rtsp-server/rtsp-session.h:
4313 rtsp-session: Use monotonic time for RTSP session timeout
4314 Changed RTSP session timeout handling to monotonic time
4315 and deprecating the API for current system time.
4316 This fixes timeouts when the system time changes.
4317 https://bugzilla.gnome.org/show_bug.cgi?id=743346
4319 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
4321 * gst/rtsp-server/rtsp-client.c:
4322 * gst/rtsp-server/rtsp-media.c:
4323 rtsp-client: Only error out in PLAY if seeking actually failed
4324 If the media was just not seekable, we continue from whatever position we are
4325 and let the client decide if that is what is wanted or not.
4326 Only if the actual seek failed, we can't really recover and should error out.
4328 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
4330 * gst/rtsp-server/rtsp-stream.c:
4331 rtsp-stream: Add necessary queues between tee and multiudpsink
4332 https://bugzilla.gnome.org/show_bug.cgi?id=744379
4334 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4336 * gst/rtsp-server/rtsp-client.c:
4337 * gst/rtsp-server/rtsp-media.c:
4338 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
4339 Instead error out properly the same way as if the SEEKING query already
4342 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
4344 * gst/rtsp-server/rtsp-stream.h:
4345 rtsp-stream: minor code formatting fix
4347 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4349 * gst/rtsp-server/rtsp-media.c:
4350 rtsp-media: fix logic for collect_streams
4351 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
4352 all streams it knows if it got any, and can check if the transport mode is OK.
4355 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4357 * gst/rtsp-server/rtsp-media.c:
4358 rtsp-media: Don't set the transport mode based on what elements we find
4359 Just print a warning if the one that was set before disagrees with what
4360 elements we found. It must already be set to something before as this
4361 function is called after we received the SDP from ANNOUNCE in RECORD mode,
4362 and we would reject ANNOUNCE if the RECORD flag was not set.
4364 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4366 * tests/check/gst/rtspserver.c:
4367 tests: rtspserver: rename shadowed variable
4368 We have two different 'sink' variables here,
4369 rename one of them for clarity.
4371 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4373 * gst/rtsp-server/rtsp-client.c:
4374 rtsp-client: fix awkward if clause
4376 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
4378 * examples/test-uri.c:
4379 examples: test-uri: improve uri argument handling and accept file names
4380 Print an error if the argument passed is not a URI and can't
4381 be converted into one, or no arguments have been provided.
4383 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4385 * examples/test-uri.c:
4386 examples: test-uri: don't remove mount point after 10 seconds
4387 It's very irritating when trying to test stuff repeatedly
4388 and serves no real purpose other than showing that it can
4391 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4393 * examples/.gitignore:
4394 examples: add new test-record to .gitignore
4396 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4398 * examples/test-record.c:
4399 * gst/rtsp-server/rtsp-client.c:
4400 * gst/rtsp-server/rtsp-media-factory.c:
4401 * gst/rtsp-server/rtsp-media-factory.h:
4402 * gst/rtsp-server/rtsp-media.c:
4403 * gst/rtsp-server/rtsp-media.h:
4404 * tests/check/gst/rtspserver.c:
4405 rtsp-media: Use flags to distinguish between PLAY and RECORD media
4407 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
4409 * examples/test-record.c:
4410 test-record: Set latency for playback-style example to 2s instead of 200ms
4412 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4414 * tests/check/gst/rtspserver.c:
4415 tests: add some unit tests for ANNOUNCE and RECORD
4416 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4418 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
4420 * gst/rtsp-server/rtsp-client.c:
4421 rtsp-client: fix a couple of leaks in handle_announce
4423 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
4425 * gst/rtsp-server/rtsp-media-factory.c:
4426 * gst/rtsp-server/rtsp-media-factory.h:
4427 * gst/rtsp-server/rtsp-media.c:
4428 * gst/rtsp-server/rtsp-media.h:
4429 rtsp-media: Expose latency setting for setting the rtpbin latency
4431 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4433 * examples/test-record.c:
4434 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
4436 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
4438 * gst/rtsp-server/rtsp-stream.c:
4439 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
4441 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
4443 * examples/Makefile.am:
4444 * examples/test-record.c:
4445 * gst/rtsp-server/rtsp-client.c:
4446 * gst/rtsp-server/rtsp-client.h:
4447 * gst/rtsp-server/rtsp-media-factory.c:
4448 * gst/rtsp-server/rtsp-media-factory.h:
4449 * gst/rtsp-server/rtsp-media.c:
4450 * gst/rtsp-server/rtsp-media.h:
4451 * gst/rtsp-server/rtsp-session-media.c:
4452 * gst/rtsp-server/rtsp-stream.c:
4453 * gst/rtsp-server/rtsp-stream.h:
4454 Add initial support for RECORD
4455 We currently only support media that is RECORD or PLAY only, not both at once.
4456 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4458 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
4460 * gst/rtsp-server/rtsp-stream.c:
4461 rtsp-stream: RTCP and RTP transport cache cookies seperated
4462 RTCP packets were not sent because the same tr_cache_cookie was used for
4463 both RTP and RTCP. So only one of the tr_cache lists were populated
4464 depending on which one was sent first. If the tr_cache list is not
4465 populated then no packets can be sent. Most often this happened to be
4466 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
4467 resulted in both the tr_cache_lists to be populated regardless of which
4469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
4471 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
4473 * gst/rtsp-server/rtsp-stream.c:
4474 rtsp-stream: fix false compiler warning
4475 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
4477 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
4479 * gst/rtsp-server/rtsp-client.c:
4480 rtsp-client: log interleaved data received
4482 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
4484 * gst/rtsp-server/rtsp-client.c:
4485 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
4487 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4489 * gst/rtsp-server/rtsp-client.c:
4490 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
4492 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4494 * gst/rtsp-server/rtsp-client.c:
4495 rtsp-client: Use a random session ID in the SDP
4496 RFC4566 Section 5.2 says that it should make the username, session id,
4497 nettype, addrtype and unicast address tuple globally unique. Always using
4498 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
4499 Instead let's create a 64 bit random number, which at least brings us
4500 closer to the goal of global uniqueness.
4501 https://tools.ietf.org/html/rfc4566#section-5.2
4503 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4505 * examples/test-launch.c:
4506 * examples/test-mp4.c:
4507 * examples/test-ogg.c:
4508 * examples/test-uri.c:
4509 examples: Don't call gst_init() and gst_get_option_group()
4510 The latter calls the former at the appropriate time.
4512 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4514 * gst/rtsp-server/rtsp-client.c:
4515 rtsp-client: Drop trailing \0 of RTSP DATA messages
4516 We add a trailing \0 in GstRTSPConnection to make parsing of
4517 string message bodies easier (e.g. the SDP from DESCRIBE) but
4518 for actual data this means we have to drop it or otherwise
4519 create invalid data.
4521 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
4523 * gst/rtsp-server/rtsp-stream.c:
4524 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
4525 Fixes crash when two threads access handle_new_sample() at the same
4526 time, one for RTP, one for RTCP.
4527 Otherwise, when iterating over the transports cache, it might be modified by
4528 another thread at the same time if the transports cookie has changed.
4529 https://bugzilla.gnome.org/show_bug.cgi?id=742954
4531 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4533 * gst/rtsp-server/rtsp-stream.c:
4534 rtsp-stream: Set format=TIME on our app sources for TCP
4536 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
4538 * gst/rtsp-server/rtsp-session-pool.c:
4539 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
4540 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
4541 RFC 2326 states that session IDs may consist of alphanumeric as well as
4542 the safe characters $-_.+ -- N.B. the percent character is not allowed.
4543 Previously the session ID was URI-escaped, this meant that any character
4544 which was not alphanumeric or any of the characters +-._~ would be
4545 percent encoded. While the RFC (surprisingly) mentions that linear white
4546 space in session IDs should be URI-escaped, it does not say anything
4547 about other characters. Moreover no white space is allowed in the
4548 session ID. Finally the percent character which is the result of
4549 URI-escaping is not allowed in a session ID.
4550 So there is no reason to do any URI-escaping, and now it is removed.
4551 https://bugzilla.gnome.org/show_bug.cgi?id=742869
4553 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
4556 Automatic update of common submodule
4557 From f2c6b95 to bc76a8b
4559 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
4562 Fix 'make check' from top-level directory
4564 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4566 * examples/test-launch.c:
4567 * examples/test-mp4.c:
4568 * examples/test-ogg.c:
4569 * examples/test-uri.c:
4570 examples: Add command-line parsing and take a 'port' argument
4571 This allows users to run multiple servers on different ports for testing.
4572 Only done for examples that actually take arguments and hence are capable of
4573 outputting different streams for each instance on each port.
4574 https://bugzilla.gnome.org/show_bug.cgi?id=742115
4576 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4578 * gst/rtsp-server/rtsp-client.c:
4579 * gst/rtsp-server/rtsp-client.h:
4580 rtsp-client: Add a send_message default signal handler
4581 This allows subclasses to easily hook into the response sending
4582 mechanism without doing everything from a signal, which seems
4583 awkward from subclasses.
4585 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4588 Automatic update of common submodule
4589 From ef1ffdc to f2c6b95
4591 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4595 configure: add --disable-examples switch
4596 https://bugzilla.gnome.org/show_bug.cgi?id=741678
4598 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
4600 * examples/.gitignore:
4601 * examples/Makefile.am:
4602 * examples/test-video-rtx.c:
4603 examples: add a retransmisison example implementing RFC4588
4604 Currently only SSRC-multiplexed rtx streams are supported
4606 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
4608 * gst/rtsp-server/rtsp-stream.c:
4609 rtsp-stream: Fix some minor memory leaks
4611 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
4613 * gst/rtsp-server/rtsp-media.c:
4614 rtsp-media: Some minor cleanup
4616 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4618 * gst/rtsp-server/rtsp-stream.c:
4619 rtsp-stream: Fix compiler warnings
4620 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
4621 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4623 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
4624 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4627 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
4629 * docs/libs/gst-rtsp-server-sections.txt:
4630 * gst/rtsp-server/rtsp-media-factory.c:
4631 * gst/rtsp-server/rtsp-media-factory.h:
4632 * gst/rtsp-server/rtsp-media.c:
4633 * gst/rtsp-server/rtsp-media.h:
4634 * gst/rtsp-server/rtsp-sdp.c:
4635 * gst/rtsp-server/rtsp-stream.c:
4636 * gst/rtsp-server/rtsp-stream.h:
4637 media: implement ssrc-multiplexed retransmission support
4638 based off RFC 4588 and the server-rtpaux example in -good
4640 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
4642 * gst/rtsp-server/rtsp-client.c:
4643 * gst/rtsp-server/rtsp-stream-transport.c:
4644 * gst/rtsp-server/rtsp-stream.c:
4645 rtsp: Ref transports in hash table.
4646 Also ref streams for transports.
4647 This solves a crash when reciving a rtcp after teardown but before
4649 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
4651 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
4654 Automatic update of common submodule
4655 From 7bb2bce to ef1ffdc
4657 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
4659 * gst/rtsp-server/rtsp-client.c:
4660 client: refactor cleanup of cached media
4662 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
4664 * tests/check/gst/client.c:
4666 The session leak is now fixed, lets remove those FIXME comments.
4668 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
4670 * tests/check/gst/rtspserver.c:
4671 tests: Test to setup two sessions on one connection
4672 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4674 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
4676 * tests/check/gst/rtspserver.c:
4677 tests: Test setup with tcp transport
4678 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4680 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
4682 * gst/rtsp-server/rtsp-client.c:
4683 client: Configure transport after creating session media
4684 The default implementation of configure_client_transport() in
4685 rtsp-client uses the session media when it chooses channels for
4686 interleaved traffic.
4687 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4689 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
4691 * gst/rtsp-server/rtsp-client.c:
4692 * gst/rtsp-server/rtsp-session-media.c:
4693 client: Stop caching media in client when doing setup
4694 If the media has been managed by a session media, it should not be
4695 cached in the client any longer. The GstRTSPSessionMedia object is now
4696 responsible for unpreparing the GstRTSPMedia object using
4697 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
4699 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4701 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4703 * gst/rtsp-server/rtsp-stream.c:
4704 rtsp-stream: unref srtp decoder when leaving bin
4705 https://bugzilla.gnome.org/show_bug.cgi?id=739481
4707 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4709 * gst/rtsp-server/rtsp-client.c:
4710 rtsp-client: mikey memory leaks
4711 https://bugzilla.gnome.org/show_bug.cgi?id=739383
4713 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
4716 Automatic update of common submodule
4717 From 84d06cd to 7bb2bce
4719 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
4722 Parallelise 'make check-valgrind'
4724 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
4727 Automatic update of common submodule
4728 From a8c8939 to 84d06cd
4730 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
4733 Automatic update of common submodule
4734 From 36388a1 to a8c8939
4736 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4738 * gst/rtsp-server/rtsp-media.c:
4739 rtsp-media: deactivate media when shutting down from paused
4740 This was only done when going directly from playing.
4741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
4743 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4745 * gst/rtsp-server/rtsp-client.c:
4746 * gst/rtsp-server/rtsp-context.h:
4747 rtsp-client: add stream transport to context
4748 We add the stream transport to the context so we can get the configured
4749 client stream transport in the setup request signal.
4750 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
4752 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4754 * gst/rtsp-server/rtsp-stream.c:
4755 stream: release lock even not all transports have been removed
4756 We don't want to keep the lock even we return FALSE because not all the
4757 transports have been removed. This could lead into a deadlock.
4758 https://bugzilla.gnome.org/show_bug.cgi?id=737797
4760 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
4762 * gst/rtsp-server/rtsp-sdp.c:
4763 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
4764 These were renamed in GstRTPBasePayload in 1.0
4766 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4768 * gst/rtsp-server/rtsp-client.c:
4769 client: set session media to NULL without the lock
4770 We need to set session medias to NULL without the client lock otherwise
4771 we can end up in a deadlock if another thread is waiting for the lock
4772 and media unprepare is also waiting for that thread to end.
4773 https://bugzilla.gnome.org/show_bug.cgi?id=737690
4775 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
4777 * gst/rtsp-server/rtsp-media.c:
4778 rtsp-media: Set state to UNPREPARING in all cases
4780 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
4782 * gst/rtsp-server/rtsp-media.c:
4783 media: set state to unpreparing when unprepare is initiated
4784 https://bugzilla.gnome.org/show_bug.cgi?id=737675
4786 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
4788 * gst/rtsp-server/rtsp-client.c:
4789 rtsp-client: Remove backlog limit while processings requests
4790 If the backlog limit is kept two cases of deadlocks may be
4791 encountered when streaming over TCP. Without the backlog
4792 limit this deadlocks can not happen, at the expence of
4794 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
4796 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
4798 * gst/rtsp-server/rtsp-client.c:
4799 rtsp-client: do not free main context before rtsp watch
4800 https://bugzilla.gnome.org/show_bug.cgi?id=737110
4802 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
4804 * tests/check/gst/rtspserver.c:
4805 tests: Extend unit test timeout to accomodate for valgrind
4806 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4808 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
4810 * gst/rtsp-server/rtsp-client.c:
4811 * gst/rtsp-server/rtsp-session.c:
4812 * gst/rtsp-server/rtsp-stream-transport.c:
4813 rtsp-*: Treat sending packets to clients as keepalive
4814 As long as gst-rtsp-server can successfully send RTP/RTCP data to
4815 clients then the client must be reading. This change makes the server
4816 timeout the connection if the client stops reading.
4817 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4819 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
4821 * gst/rtsp-server/rtsp-client.c:
4822 rtsp-client: Allow backlog to grow while expiring session
4823 Allow the send backlog in the RTSP watch to grow to unlimited size while
4824 attempting to bring the media pipeline to NULL due to a session
4825 expiring. Without this change the appsink element cannot change state
4826 because it is blocked while rendering data in the new_sample callback.
4827 This callback will block until it has successfully put the data into the
4828 send backlog. There is a chance that the send backlog is full at this
4829 point which means that the callback may block for a long time, possibly
4830 forever. Therefore the media pipeline may also be prevented from
4831 changing state for a long time.
4832 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4834 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
4836 * gst/rtsp-server/rtsp-client.c:
4837 rtsp-client: Make old compilers happy
4838 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
4839 Just in case that guint8 doesn't fit in a pointer. Just in case ...
4841 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
4843 * gst/rtsp-server/rtsp-client.c:
4844 client: raise the backlog limits before pausing
4845 We need to raise the backlog limits before pausing the pipeline or else
4846 the appsink might be blocking in the render method in wait_backlog() and
4847 we would deadlock waiting for paused.
4848 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
4850 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
4852 * gst/rtsp-server/rtsp-client.c:
4853 client: make define for the WATCH_BACKLOG
4854 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
4856 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
4858 * gst/rtsp-server/rtsp-client.c:
4859 client: simplify session transport handling
4860 link/unlink of the transport in a session was done to keep track of all
4861 TCP transports and to send RTP/RTCP data to the streams. We can simplify
4862 that by putting all the TCP transports in a hashtable indexed with the
4864 We also don't need to link/unlink the transports when we pause/resume
4865 the streams. The same effect is already achieved when we pause/play the
4866 media. Indeed, when we pause the media, the transport is removed from
4867 the media and the callbacks will not be called anymore.
4868 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
4870 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
4872 * gst/rtsp-server/rtsp-stream-transport.c:
4873 * gst/rtsp-server/rtsp-stream-transport.h:
4874 stream-transport: make method to handle received data
4875 Make a method to handle the data received on a channel. It sends the
4876 data to the stream of the transport on the RTP or RTCP pads based on
4879 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
4881 * examples/test-mp4.c:
4882 test: add example of dumping RTCP reports
4884 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
4886 * gst/rtsp-server/rtsp-media.c:
4887 * gst/rtsp-server/rtsp-stream.c:
4888 * gst/rtsp-server/rtsp-stream.h:
4889 rtsp-media: Make sure that sequence numbers are monotonic after pause
4890 The sequence number is not monotonic for RTP packets after pause. The
4891 reason is basepayloader generates a randon sequence number when the
4892 pipeline goes from ready to pause. With this fix generation of sequence
4893 number will be monotonic when going from pause to play request.
4894 https://bugzilla.gnome.org/show_bug.cgi?id=736017
4896 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
4898 * gst/rtsp-server/rtsp-client.c:
4899 rtsp-client: Protect saved clients watch with a mutex
4900 Fixes a crash when close() is called while merging clients
4901 in handle_tunnel(). In that case close() would destroy the
4902 watch while it is still being used in handle_tunnel().
4903 https://bugzilla.gnome.org/show_bug.cgi?id=735570
4905 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
4907 * gst/rtsp-server/rtsp-stream.c:
4908 rtsp-stream: Remove the multicast group udp sources when removing from the bin
4910 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4912 * gst/rtsp-server/rtsp-media.c:
4913 * gst/rtsp-server/rtsp-stream.c:
4914 * gst/rtsp-server/rtsp-stream.h:
4915 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
4916 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
4917 seeking and will always continue counting the time. This leads to
4918 the NPT after a backwards seek to be something completely different
4919 to the actual seek position.
4920 https://bugzilla.gnome.org/show_bug.cgi?id=732644
4922 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
4924 * examples/test-appsrc.c:
4925 examples: fix another reference leak
4926 gst_rtsp_media_get_element() returns a new ref.
4928 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4930 * examples/test-appsrc.c:
4931 examples: unref element after usage
4932 gst_bin_get_by_name_recurse_up() returns an element
4933 reference that must be unreffed after usage.
4934 https://bugzilla.gnome.org/show_bug.cgi?id=734546
4936 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
4938 * gst/rtsp-server/rtsp-media.c:
4939 signals: Fix copy-pasto in target-state signal offset
4941 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
4945 Makefile: Add usage of build-checks step
4946 Allows building checks without running them
4948 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
4950 * gst/rtsp-server/rtsp-stream.c:
4951 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
4952 When a UDP multicast transport is used it is expected that the server listens
4953 for RTP and RTCP packets on the multicast group with the corresponding port.
4954 Without this we will never get RTCP packets from clients in multicast mode.
4955 https://bugzilla.gnome.org/show_bug.cgi?id=732238
4957 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
4962 === release 1.4.0 ===
4964 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
4970 * gst-rtsp-server.doap:
4973 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
4975 * gst/rtsp-server/rtsp-media.h:
4976 media: correct misspelled words in description
4977 https://bugzilla.gnome.org/show_bug.cgi?id=733244
4979 === release 1.3.91 ===
4981 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4987 * gst-rtsp-server.doap:
4990 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
4992 * docs/libs/gst-rtsp-server-sections.txt:
4995 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
4997 * gst/rtsp-server/rtsp-server.c:
4998 server: implement client REMOVE filter
5000 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
5002 * gst/rtsp-server/rtsp-client.c:
5003 * gst/rtsp-server/rtsp-client.h:
5004 client: expose _close() method
5005 Expose a previously internal close method to close the client
5008 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
5010 * gst/rtsp-server/rtsp-session-pool.c:
5011 session-pool: signal session-removed outside of the lock
5012 Release the lock before emiting the session-removed signal.
5014 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
5016 * gst/rtsp-server/rtsp-client.c:
5017 * gst/rtsp-server/rtsp-server.c:
5018 * gst/rtsp-server/rtsp-session-pool.c:
5019 * gst/rtsp-server/rtsp-session.c:
5020 * gst/rtsp-server/rtsp-stream.c:
5021 filter: Release lock in filter functions
5022 Release the object lock before calling the filter functions. We need to
5023 keep a cookie to detect when the list changed during the filter
5024 callback. We also keep a hashtable to make sure we only call the filter
5025 function once for each object in case of concurrent modification.
5026 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
5028 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
5030 * gst/rtsp-server/rtsp-client.c:
5031 client: check if watch is set in handle_teardown()
5032 The unit tests run without a watch
5034 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5036 * tests/check/gst/client.c:
5037 client tests: send teardown to cleanup session
5039 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
5041 * tests/check/gst/rtspserver.c:
5042 server tests: send teardown to cleanup session
5044 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5046 * gst/rtsp-server/rtsp-client.c:
5047 client: keep ref to client for the session removed handler
5048 This extra ref will be dropped when all client sessions have been
5049 removed. A session is removed when a client sends teardown, closes its
5050 endpoint of the TCP connection or the sessions expires.
5051 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5053 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
5055 * gst/rtsp-server/rtsp-client.c:
5056 * gst/rtsp-server/rtsp-session.c:
5057 * tests/check/gst/client.c:
5058 client: manage media in session as a last step
5059 Once we manage a media in a session, we can't unmanage it anymore
5060 without destroying it. Therefore, first check everything before we
5061 manage the media, otherwise if something is wrong we have no way to
5063 If we created a new session and something went wrong, remove the session
5064 again. Fixes a leak in the unit test.
5066 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5068 * examples/test-mp4.c:
5069 * examples/test-ogg.c:
5070 examples: print 'stream ready at url' for mp4 and ogg example
5072 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
5074 * gst/rtsp-server/rtsp-client.c:
5075 * gst/rtsp-server/rtsp-sdp.c:
5076 rtsp: fix for MIKEY api change
5078 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
5080 * gst/rtsp-server/rtsp-client.c:
5081 client: free watch context only once
5082 The watch context is freed when the source is destroyed. Avoids
5083 a CRITICAL when we try to unref the context twice.
5085 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
5087 * gst/rtsp-server/rtsp-client.c:
5090 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
5092 * gst/rtsp-server/rtsp-client.c:
5093 client: protect sessions with lock
5094 Protect the list of sessions with the lock.
5095 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5097 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
5099 * gst/rtsp-server/rtsp-client.c:
5100 Client: keep a ref to the session
5101 Don't just keep a weak ref to the session objects but use a hard ref. We
5102 will be notified when a session is removed from the pool (expired) with
5103 the new session-removed signal.
5104 Don't automatically close the RTSP connection when all the sessions of
5105 a client are removed, a client can continue to operate and it can create
5106 a new session if it wants. If you want to remove the client from the
5107 server, you have to use gst_rtsp_server_client_filter() now.
5108 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
5109 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
5111 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
5113 * gst/rtsp-server/rtsp-session-pool.c:
5114 * gst/rtsp-server/rtsp-session-pool.h:
5115 session-pool: add session-removed signal
5116 Add a signal to be notified when a session is removed from the pool.
5118 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
5120 * gst/rtsp-server/Makefile.am:
5121 * gst/rtsp-server/rtsp-server.h:
5122 Make rtsp-server.h a single-include header, use it for G-I
5123 https://bugzilla.gnome.org/show_bug.cgi?id=732411
5125 === release 1.3.90 ===
5127 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
5133 * gst-rtsp-server.doap:
5136 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
5138 * gst/rtsp-server/rtsp-stream.c:
5139 stream: crypto can be NULL
5141 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
5143 * gst/rtsp-server/rtsp-client.c:
5144 * gst/rtsp-server/rtsp-media.c:
5145 * gst/rtsp-server/rtsp-mount-points.c:
5146 introspection: add missing allow-none annotations
5147 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5149 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
5151 * gst/rtsp-server/rtsp-address-pool.c:
5152 * gst/rtsp-server/rtsp-media.c:
5153 * gst/rtsp-server/rtsp-session-media.c:
5154 * gst/rtsp-server/rtsp-session-pool.c:
5155 * gst/rtsp-server/rtsp-stream-transport.c:
5156 * gst/rtsp-server/rtsp-stream.c:
5157 * gst/rtsp-server/rtsp-token.c:
5158 introspection: add (nullable) annotations to return values
5159 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5161 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
5163 * gst/rtsp-server/rtsp-client.c:
5164 * gst/rtsp-server/rtsp-stream.c:
5165 gi: improve annotations
5166 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
5168 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
5170 * gst/rtsp-server/rtsp-client.c:
5171 * gst/rtsp-server/rtsp-media-factory.c:
5172 * gst/rtsp-server/rtsp-media.c:
5173 * gst/rtsp-server/rtsp-server.c:
5174 signals: use generic marshal function
5175 Use the generic C marshal function.
5176 Use more explicit type instead of G_TYPE_POINTER
5178 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
5180 * gst/rtsp-server/rtsp-context.h:
5181 context: add type macro
5183 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
5185 * gst/rtsp-server/rtsp-client.c:
5186 * gst/rtsp-server/rtsp-sdp.c:
5187 * gst/rtsp-server/rtsp-sdp.h:
5188 sdp: hide key length defines
5189 They don't have a namespace.
5191 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5196 === release 1.3.3 ===
5198 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
5204 * gst-rtsp-server.doap:
5207 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5209 * gst/rtsp-server/rtsp-client.c:
5210 * gst/rtsp-server/rtsp-sdp.c:
5211 * gst/rtsp-server/rtsp-sdp.h:
5212 mikey: add different key length parameters
5213 Add encryption and authentication key length parameters to MIKEY. For
5214 the encoders, the key lengths are obtained from the cipher and auth
5215 algorithms set in the caps. For the decoders, they are obtained while
5216 parsing the key management from the client.
5217 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
5219 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
5221 * tests/check/gst/stream.c:
5222 stream tests: Make sure we get right multicast address from stream
5223 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
5225 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5227 * gst/rtsp-server/rtsp-client.c:
5228 client: ref the context until rtsp watch is alive
5229 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
5231 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5233 * gst/rtsp-server/rtsp-client.c:
5234 client: Destroy the rtsp watch after connection close
5236 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
5238 * gst/rtsp-server/rtsp-media.c:
5239 media: fix confusing comment
5241 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
5243 * gst/rtsp-server/rtsp-session.c:
5244 rtsp-session: Timeout in header.
5245 Adding the possbilty to always have timout in header.
5246 This is configurabe with setting "timeout-always-visible".
5247 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
5249 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
5254 === release 1.3.2 ===
5256 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
5263 * gst-rtsp-server.doap:
5266 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5269 Automatic update of common submodule
5270 From 211fa5f to 1f5d3c3
5272 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
5274 * gst/rtsp-server/rtsp-client.c:
5275 client: store TCP ports in transport
5276 Store the TCP ports in the transport when we are doing RTSP over TCP.
5277 This way, we can easily get to the ports from the transport.
5278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
5280 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5282 * gst/rtsp-server/rtsp-stream.c:
5283 stream: add signals for new RTP/RTCP encoders
5284 New signals to allow the user to configure the dynamically created
5286 https://bugzilla.gnome.org/show_bug.cgi?id=730228
5288 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5290 * gst/rtsp-server/rtsp-media.c:
5291 * gst/rtsp-server/rtsp-media.h:
5292 media: Make suspend()/unsuspend() virtual
5293 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
5295 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5297 * gst/rtsp-server/rtsp-client.c:
5298 client: fix send-message signal marshaller
5299 Use generic marshalling for the send-message signal. It has
5300 two POINTER arguments, not just one.
5301 https://bugzilla.gnome.org/show_bug.cgi?id=729900
5303 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
5305 * tests/check/gst/media.c:
5306 tests: add and remove pads only once
5307 In this test we simulate a dynamic pad by watching the caps event.
5308 Because of renegotiation in the base payloader now, this caps is sent
5309 multiple times but we can only deal with 1 invocation, use a variable to
5310 only 'add and remove' the pad once.
5312 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
5314 * tests/check/gst/rtspserver.c:
5315 tests: add unit test for correct handling of Require headers
5316 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5318 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5320 * gst/rtsp-server/rtsp-client.c:
5321 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
5322 Servers must handle Require headers and must report a failure
5323 if they don't handle any of the Required options, see RFC 2326,
5324 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
5325 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5327 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5332 === release 1.3.1 ===
5334 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5340 * gst-rtsp-server.doap:
5343 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
5346 Automatic update of common submodule
5347 From bcb1518 to 211fa5f
5349 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
5354 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5356 * tests/check/gst/sessionmedia.c:
5357 tests: fix memory leak in sessionmedia unit test
5359 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
5361 * gst/rtsp-server/rtsp-client.c:
5362 client: emit a signal before sending a message
5363 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
5365 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
5367 * gst/rtsp-server/rtsp-client.c:
5368 client: pass context to send_message
5369 Pass the current context to send_message, we will need it later.
5371 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
5373 * gst/rtsp-server/rtsp-client.c:
5374 client: fix typo in comment
5376 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
5378 * gst/rtsp-server/rtsp-media.c:
5379 media: Do not stop thread twice if default_prepare() fails
5381 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
5383 * gst/rtsp-server/rtsp-client.c:
5384 client: set the watch to flushing before going to NULL
5385 First set the watch to flushing so that we unblock any current and
5386 future attempt to send data on the watch, Then set the pipeline to
5388 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
5390 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
5392 * gst/rtsp-server/rtsp-session-pool.c:
5393 * tests/check/gst/sessionpool.c:
5394 rtsp-session-pool: Fixes annotation
5395 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
5396 in the sessionpool test.
5397 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
5399 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
5401 * gst/rtsp-server/rtsp-media.c:
5402 * gst/rtsp-server/rtsp-media.h:
5403 media: make media_prepare virtual
5404 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
5406 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5408 * gst/rtsp-server/rtsp-media.c:
5409 * tests/check/gst/media.c:
5410 media: stop the thread in more error cases
5412 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5414 * gst/rtsp-server/rtsp-media.c:
5415 * tests/check/gst/media.c:
5416 media: allow NULL as the thread
5417 Use the default context whan passing a NULL thread.
5419 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5421 * gst/rtsp-server/rtsp-client.c:
5422 rtsp-client: indent cleanup
5423 Coverity was moaning about unreachable code, and I think it was just
5424 confused by { being before the label. We'll see if it pops up again.
5427 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
5429 * gst/rtsp-server/rtsp-client.c:
5430 * gst/rtsp-server/rtsp-media.c:
5431 client: Add drop-backlog property
5432 When we have too many messages queued for a client (currently hardcoded
5433 to 100) we overflow and drop the messages. Add a drop-backlog property
5434 to control this behaviour. Setting this property to FALSE will retry
5435 to send the messages to the client by waiting for more room in the
5437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
5439 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
5441 * gst/rtsp-server/rtsp-client.c:
5442 client: support for POST before GET when setting up a tunnel
5444 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
5446 * gst/rtsp-server/rtsp-client.c:
5447 client: remove watch of the second client after http tunnel setup
5448 The second client will be freed after the HTTP tunnel has been set up.
5449 Make sure it's RTSP watch is never dispatched again.
5450 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
5452 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
5454 * gst/rtsp-server/rtsp-media.c:
5455 * tests/check/gst/media.c:
5456 media: Make media_prepare() fail if port allocation fails
5457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
5459 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
5461 * tests/check/gst/media.c:
5462 media test: cleanup the thread pool in tests
5464 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
5466 * gst/rtsp-server/rtsp-media.c:
5467 * tests/check/gst/media.c:
5468 rtsp-media: Unblock blocked streams in unprepare
5469 The streams will be blocked when a live media is prepared.
5470 The streams should be unblocked in gst_rtsp_media_unprepare.
5471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
5473 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
5475 * gst/rtsp-server/rtsp-media.c:
5476 media: release the state lock when going to NULL
5477 Set our state to UNPREPARING and release the state-lock before
5478 setting the pipeline to the NULL state. This way, any pad-added
5479 callback will be able to take the state-lock and check that we are now
5480 unpreparing instead of deadlocking.
5481 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
5483 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
5485 * gst/rtsp-server/rtsp-media.c:
5486 media: protect status with lock
5487 Make sure we only update the status with the lock.
5489 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
5491 * gst/rtsp-server/rtsp-client.c:
5492 * gst/rtsp-server/rtsp-sdp.c:
5493 rtsp: update for MIKEY API changes
5495 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
5497 * gst/rtsp-server/rtsp-client.c:
5498 client: parse the mikey response from the client
5499 Parse the mikey response from the client and update the policy for
5502 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
5504 * gst/rtsp-server/rtsp-stream.c:
5505 * gst/rtsp-server/rtsp-stream.h:
5506 stream: add method to set crypto info
5507 Make a method to configure the crypto information of a stream.
5508 Set udpsrc in READY instead of PAUSED so that we can configure caps
5511 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
5513 * gst/rtsp-server/rtsp-client.c:
5514 client: cleanup error paths
5516 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
5518 * gst/rtsp-server/rtsp-media.c:
5521 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
5523 * examples/test-video.c:
5524 test: enable SRTP only on RTSPS
5525 We only want to enable SRTP when doing rtsp over TLS so that we can
5526 exchange the keys in a secure way.
5528 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
5530 * examples/test-video.c:
5531 test: print an error on failure
5533 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
5536 * examples/test-video.c:
5537 * gst/rtsp-server/rtsp-sdp.c:
5538 * gst/rtsp-server/rtsp-stream.c:
5539 * tests/check/Makefile.am:
5540 stream: add SRTP support
5541 Install srtp encoder and decoder elements in rtpbin
5544 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5546 * tests/check/Makefile.am:
5547 * tests/check/gst/sessionpool.c:
5548 tests: Add unit tests for sessionpool
5549 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
5551 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5553 * tests/check/gst/threadpool.c:
5554 tests: Improve code coverage of rtsp-threadpool tests
5555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
5557 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5559 * tests/check/gst/sessionmedia.c:
5560 tests: Improve code coverage for rtsp-session-media
5561 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
5563 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5565 gobject-introspection: Add annotations to support language bindings
5566 In addition a few cosmetic changes:
5567 * Adjust the order of arguments
5568 * Fix typo: occured -> occurred
5569 * Fix indentation after Return:-clauses
5570 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
5572 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5574 * gst/rtsp-server/rtsp-stream.c:
5575 rtsp-stream: Don't mix IPv4 and IPv6 addresses
5576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
5578 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
5580 * gst/rtsp-server/rtsp-stream.c:
5581 stream: take caps after the session manager
5582 Take the caps for the SDP after they leave the rtpbin so that we can
5583 also get the properties added by rtpbin elements.
5585 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
5587 * gst/rtsp-server/rtsp-stream.c:
5588 stream: release lock while pushing out packets
5589 Keep a cache of the transports and use this to iterate the transport
5590 while pushing packets. This allows us to release the lock early.
5591 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
5593 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
5595 * gst/rtsp-server/rtsp-client.c:
5596 * gst/rtsp-server/rtsp-client.h:
5597 rtsp-client: vmethod for modifying tunnel GET response
5598 Add a vmethod tunnel_http_response where the response to the HTTP GET
5599 for tunneled connections can be modified.
5600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
5602 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
5604 * gst/rtsp-server/rtsp-sdp.c:
5605 sdp: make 1 media line per profile
5606 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
5607 line in the SDP for each profile. The client is then supposed to pick
5608 one of the profiles in the SETUP request. Because the m= lines have the
5609 same pt, the client also knows that only 1 option is possible.
5611 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
5613 * gst/rtsp-server/rtsp-media-factory.c:
5614 * gst/rtsp-server/rtsp-media-factory.h:
5615 * gst/rtsp-server/rtsp-media.c:
5616 factory: add profile property and pass to media and streams
5618 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
5620 * examples/test-multicast.c:
5621 * gst/rtsp-server/rtsp-sdp.c:
5622 sdp: pass multicast connection for multicast-only stream
5623 Pass the multicast address of the stream in the connection info in the
5624 SDP so that clients try a multicast connection first.
5625 Only allow multicast connections in the test-multicast example. Also
5626 increase the TTL a little.
5628 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5631 .gitignore: Ignore gcov intermediate files
5632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
5634 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
5636 * gst/rtsp-server/rtsp-stream.c:
5637 stream: release some locks in error cases
5639 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5641 docs: Enable and fix gtk-doc warnings
5642 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
5643 * addresspool/mediafactory: Add missing annotation colon
5644 * stream: Annotate return value
5645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
5647 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
5650 Automatic update of common submodule
5651 From fe1672e to bcb1518
5653 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
5656 Automatic update of common submodule
5657 From 1a07da9 to fe1672e
5659 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
5661 * examples/Makefile.am:
5662 examples: use LDADD for libs instead of LDFLAGS
5664 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
5667 configure: make sure releases are in .doap file
5669 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
5671 * examples/test-cgroups.c:
5672 examples: test-cgroups: don't put code with side effects into g_assert()
5673 The g_assert() might get compiled out with the right
5674 compiler/preprocessor flags.
5676 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
5678 * examples/.gitignore:
5679 examples: add cgroup test binary to .gitignore
5681 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
5683 * examples/test-cgroups.c:
5684 examples: fix cgroup test build
5685 Fixes build failure caused by compiler warning:
5686 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
5688 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
5691 .gitignore: ignore temp files created in the course of 'make check'
5693 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
5695 * gst/rtsp-server/rtsp-media.c:
5696 rtsp-media: don't loose frames handling new PLAY request
5697 If client supplied a range check if the range specifies the start point.
5698 If not, then do an accurate seek to the current position. If a start
5699 point was specified do do a key unit seek to make sure the streaming
5700 starts with decodeable frames.
5701 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
5703 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
5705 * gst/rtsp-server/rtsp-media.c:
5706 Revert "media: only flush when setting a new start position"
5707 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
5708 We need to do the flush in all cases, demuxer block currently for
5711 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
5713 * gst/rtsp-server/rtsp-media.c:
5714 media: only flush when setting a new start position
5715 Only flush the pipeline when we change the start position with
5717 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
5719 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
5721 * gst/rtsp-server/rtsp-stream.c:
5722 stream: set ttl-mc before adding the socket
5723 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
5724 never be set on socket.
5725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
5727 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5729 * gst/rtsp-server/rtsp-media.c:
5730 media: stop thread if media is already prepared
5731 in gst_rtsp_media_prepare() the thread is not used if media is already
5732 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
5734 https://bugzilla.gnome.org/show_bug.cgi?id=724182
5736 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
5739 build: Ship gst-rtsp-server.doap file
5741 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
5743 * tests/check/gst/rtspserver.c:
5744 tests: Fix another compiler warning with gcc
5746 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
5748 * gst/rtsp-server/rtsp-client.c:
5749 * gst/rtsp-server/rtsp-mount-points.c:
5750 * gst/rtsp-server/rtsp-stream.c:
5751 * tests/check/gst/client.c:
5752 rtsp-server: Fix lots of compiler warnings with clang
5754 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
5757 * gst-rtsp-server.doap:
5758 * tests/Makefile.am:
5759 configure: Synchronise with the configure scripts of the other modules
5761 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
5764 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
5766 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
5768 * gst/rtsp-server/rtsp-media.c:
5769 * gst/rtsp-server/rtsp-stream.c:
5770 Revert "rtsp-server: support build against last stable release"
5771 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
5772 Let us require 1.2.3 now, which is going to be released in a few
5775 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
5777 * gst/rtsp-server/rtsp-session-media.c:
5778 * gst/rtsp-server/rtsp-stream-transport.c:
5779 session: improve RTP-Info
5780 Ignore streams that can't generate RTP-Info instead of failing.
5781 Don't return the empty string when all streams are unconfigured but
5782 return NULL so that we don't generate and empty RTP-Info header.
5783 Improve docs a little.
5785 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
5787 * gst/rtsp-server/rtsp-session-media.c:
5788 Don't free rtpinfo GString when it is NULL
5789 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5791 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
5793 * gst/rtsp-server/rtsp-media.c:
5794 media: only set keyframe flag when modifying start
5795 Only set the keyframe flag when we modify the start position. The
5796 keyframe flag should probably be ignored when no change is requested but
5797 until we can claim this is all documented properly and all demuxer
5798 implement this, avoid setting the flag.
5799 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
5801 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
5803 * gst/rtsp-server/rtsp-thread-pool.c:
5804 thread-pool: Unref source after mainloop has quit to avoid races in GLib
5805 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
5807 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
5809 * gst/rtsp-server/rtsp-stream.c:
5810 stream: handle NULL seqnum and rtptime arguments
5812 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
5814 * gst/rtsp-server/rtsp-thread-pool.c:
5815 * tests/check/gst/threadpool.c:
5816 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
5817 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
5819 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
5821 * gst/rtsp-server/rtsp-stream.c:
5822 stream: add fallback for missing stats property
5823 Use a fallback when the payloader does not have a stats property
5824 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5826 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
5829 Automatic update of common submodule
5830 From f7bc1c3 to 1a07da9
5832 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
5834 * gst/rtsp-server/rtsp-stream.c:
5835 stream: don't leak stats structure
5836 Don't leak the stats structure and deal with NULL stats.
5838 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
5840 * gst/rtsp-server/rtsp-stream.c:
5841 stream: Get rtpinfo properties atomically from payloader
5842 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
5844 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
5846 * gst/rtsp-server/rtsp-media.c:
5847 media: refactor state change functions and signals
5848 Make functions to set the target state and the pipeline state and emit
5849 the signals from those functions.
5851 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
5853 * gst/rtsp-server/rtsp-media.c:
5854 * gst/rtsp-server/rtsp-media.h:
5855 media: add signal to notify of pending state changes
5857 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
5859 * gst/rtsp-server/rtsp-media.c:
5860 * gst/rtsp-server/rtsp-stream.c:
5861 rtsp-server: support build against last stable release
5862 Until 1.2.3 is out with the new get_type function and we
5865 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
5867 * gst/rtsp-server/rtsp-stream.c:
5868 stream: fix compilation
5870 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
5872 * gst/rtsp-server/rtsp-media.c:
5873 * gst/rtsp-server/rtsp-media.h:
5874 * gst/rtsp-server/rtsp-stream.c:
5875 * gst/rtsp-server/rtsp-stream.h:
5876 stream: add property to configure profiles
5878 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
5880 * gst/rtsp-server/rtsp-client.c:
5881 client: let stream check supported transport
5882 Delegate the check if a transport is allowed to the stream.
5883 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
5885 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
5887 * gst/rtsp-server/rtsp-stream.c:
5888 * gst/rtsp-server/rtsp-stream.h:
5889 stream: add method to check supported transport
5890 Add a method to check if a transport is supported
5892 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
5895 configure.ac: Only check for gstreamer-check, not check
5896 We include check in gstreamer-check since quite some time now.
5898 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
5900 * gst/rtsp-server/rtsp-session-media.c:
5901 * gst/rtsp-server/rtsp-stream-transport.c:
5902 * gst/rtsp-server/rtsp-stream.c:
5903 * gst/rtsp-server/rtsp-stream.h:
5904 stream: return clock-rate from get_rtpinfo
5905 And use it to correct the rtptime to the requested start-time.
5906 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
5908 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
5910 * gst/rtsp-server/rtsp-session-media.c:
5911 * gst/rtsp-server/rtsp-stream-transport.c:
5912 * gst/rtsp-server/rtsp-stream-transport.h:
5913 session-media: calculate start-time
5915 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
5917 * gst/rtsp-server/rtsp-stream-transport.c:
5918 * gst/rtsp-server/rtsp-stream.c:
5919 * gst/rtsp-server/rtsp-stream.h:
5920 stream: also return the running-time
5921 Return the running-time in the rtpinfo as well.
5923 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
5925 * gst/rtsp-server/rtsp-client.c:
5926 * gst/rtsp-server/rtsp-session-media.c:
5927 * gst/rtsp-server/rtsp-session-media.h:
5928 * gst/rtsp-server/rtsp-stream-transport.c:
5929 * gst/rtsp-server/rtsp-stream-transport.h:
5930 session-media: let the session-media make the RTPInfo
5931 Add method to create the RTPInfo for a stream-transport.
5932 Add method to create the RTPInfo for all stream-transports in a
5934 Use the session-media RTPInfo code in client. This allows us to refactor
5935 another method to link the TCP callbacks.
5937 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5939 mount-points: sort sequence before g_sequence_lookup
5940 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
5941 sort sequence if dirty, otherwise lookup will fail.
5942 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
5944 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
5947 configure: rename package from gst-rtsp to gst-rtsp-server
5948 To match git module name and avoid confusion with the
5949 rtsp lib in gst-plugins-base and rtsp plugin in -good.
5951 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
5954 configure: bump core/base/good requirement to 1.2.0
5955 Bump to released stable version and make implicit
5956 requirements explicit.
5958 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
5963 Fix broken gettext setup which is not used anyway
5965 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
5968 Automatic update of common submodule
5969 From dbedaa0 to d48bed3
5971 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
5973 * gst/rtsp-server/rtsp-client.c:
5974 * gst/rtsp-server/rtsp-media.c:
5975 * gst/rtsp-server/rtsp-media.h:
5976 media: add setup_sdp vmethod
5977 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
5978 gst_rtsp_media_setup_sdp.
5979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
5981 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
5983 * gst/rtsp-server/rtsp-stream.c:
5984 rtsp-stream: Check return value of sscanf
5985 streamid is only valid if sscanf matched something.
5987 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
5989 * gst/rtsp-server/rtsp-client.c:
5990 rtsp-client: Fix iteration
5991 Wouldn't even enter the code block otherwise (i++ was used as the check
5992 and not the postfix).
5994 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
5996 * gst/rtsp-server/rtsp-client.c:
5997 * gst/rtsp-server/rtsp-client.h:
5998 client: add vmethod to configure media and streams
5999 Implement a vmethod that can be used to configure the media and the
6000 streams based on the current context. Handle the blocksize handling in
6001 the default handler.
6002 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
6004 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6007 Make git ignore more unit test binaries
6009 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6011 * gst/rtsp-server/rtsp-address-pool.h:
6012 * gst/rtsp-server/rtsp-auth.h:
6013 * gst/rtsp-server/rtsp-client.h:
6014 * gst/rtsp-server/rtsp-context.h:
6015 * gst/rtsp-server/rtsp-media-factory-uri.h:
6016 * gst/rtsp-server/rtsp-media-factory.h:
6017 * gst/rtsp-server/rtsp-media.h:
6018 * gst/rtsp-server/rtsp-mount-points.h:
6019 * gst/rtsp-server/rtsp-server.h:
6020 * gst/rtsp-server/rtsp-session-media.h:
6021 * gst/rtsp-server/rtsp-session-pool.h:
6022 * gst/rtsp-server/rtsp-session.h:
6023 * gst/rtsp-server/rtsp-stream-transport.h:
6024 * gst/rtsp-server/rtsp-stream.h:
6025 * gst/rtsp-server/rtsp-thread-pool.h:
6026 * gst/rtsp-server/rtsp-token.h:
6027 rtsp-server: add padding to many public structures
6028 Not mini objects though, since they are not subclassable
6029 anyway, nor kept on the stack or inlined in a structure.
6031 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6033 media: add new create_rtpbin vmethod
6034 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
6035 https://bugzilla.gnome.org/show_bug.cgi?id=719734
6037 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
6039 * tests/check/gst/media.c:
6040 tests: fix memory leak, free test's thread pool
6041 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
6043 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
6045 * gst/rtsp-server/rtsp-stream-transport.c:
6046 stream-transport: free url in finalize
6048 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
6050 * gst/rtsp-server/rtsp-media.c:
6051 media: also do state change in suspended state
6053 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
6055 * gst/rtsp-server/rtsp-client.c:
6056 * gst/rtsp-server/rtsp-media.c:
6057 media: also handle prepare and range in suspended state
6058 When we are suspended, we are already prepared.
6059 We can get the range in the suspended state.
6061 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
6063 * tests/check/Makefile.am:
6064 * tests/check/gst/sessionmedia.c:
6065 check: add test for uri in setup
6066 Added unit tests for the new functionality in GstRTSPStreamTransport.
6067 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6069 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
6071 * gst/rtsp-server/rtsp-client.c:
6072 client: store setup uri and use in PLAY response
6073 Store the uri used when doing the setup and use that in the PLAY
6075 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6077 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
6079 * gst/rtsp-server/rtsp-stream-transport.c:
6080 * gst/rtsp-server/rtsp-stream-transport.h:
6081 stream-transport: add method to get/set url
6083 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
6085 * gst/rtsp-server/rtsp-client.c:
6086 client: suspend after SDP and unsuspend before PLAYING
6087 Based on patches by Ognyan Tonchev <ognyan@axis.com>
6088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
6090 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
6092 * gst/rtsp-server/rtsp-media-factory.c:
6093 * gst/rtsp-server/rtsp-media-factory.h:
6094 * gst/rtsp-server/rtsp-media.c:
6095 * gst/rtsp-server/rtsp-media.h:
6096 * gst/rtsp-server/rtsp-session-media.c:
6097 * gst/rtsp-server/rtsp-session.c:
6098 * tests/check/gst/media.c:
6099 * tests/check/gst/mediafactory.c:
6100 media: add suspend modes
6101 Add support for different suspend modes. The stream is suspended right after
6102 producing the SDP and after PAUSE. Different suspend modes are available that
6103 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
6104 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
6105 state and RESET will bring the pipeline to the NULL state.
6106 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
6107 this means that the pipeline needs to be prerolled again.
6108 Base on patches by Ognyan Tonchev <ognyan@axis.com>
6109 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6111 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
6113 * gst/rtsp-server/rtsp-media.c:
6114 media: start live streams in blocked state
6115 Start live streams in the blocked state and make them preroll using the
6116 messages. This ensure that no data is played by the sink until we explicitly
6117 unblock the stream right before going to PLAYING.
6118 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6120 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
6122 * gst/rtsp-server/rtsp-media.c:
6123 media: refactor starting and waiting for preroll
6124 Based on patches from Ognyan Tonchev <ognyan@axis.com>
6125 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6127 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
6129 * gst/rtsp-server/rtsp-stream.c:
6130 * gst/rtsp-server/rtsp-stream.h:
6131 stream: add API to block streams
6132 Add an API to block on the streams and make it post a message.
6133 Based on patch by Ognyan Tonchev <ognyan@axis.com>
6134 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6136 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
6138 * docs/libs/Makefile.am:
6139 docs: Specify the override file
6140 Even if it's empty (for now) it avoids make distcheck complaining
6142 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
6144 * gst/rtsp-server/rtsp-media.c:
6145 media: move default implementations to where they are used
6147 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
6149 * gst/rtsp-server/rtsp-media.c:
6150 media: take the right lock in gst_rtsp_media_set_pipeline_state()
6151 We need to take the state_lock when calling this method.
6153 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
6155 * gst/rtsp-server/rtsp-media.c:
6156 media: handle add-added on non-bins too
6157 Handle dynamic payloaders that are not bins, as used in the unit-test.
6159 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6161 * gst/rtsp-server/rtsp-media-factory.c:
6162 * gst/rtsp-server/rtsp-media-factory.h:
6163 * gst/rtsp-server/rtsp-media.c:
6164 rtsp-media/-factory: Fix request pad name comments
6165 These must be escaped for gtk-doc to parse the comments without warnings.
6167 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6169 rtsp-media: remove transports if media is in error status
6170 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
6171 trying to change to GST_STATE_NULL and media is in error status, we
6172 remove all transports.
6173 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
6175 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
6177 * gst/rtsp-server/rtsp-media.c:
6178 rtsp-media: use element metadata to find payloader
6179 Use the element metadata to find the payloader instead of checking
6181 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
6183 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6185 rtsp-stream: add getter for payload type
6186 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
6187 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
6188 element and create the stream with this one instead of the dynpay%d
6190 https://bugzilla.gnome.org/show_bug.cgi?id=712396
6192 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6194 * gst/rtsp-server/rtsp-client.c:
6195 * gst/rtsp-server/rtsp-context.h:
6196 * gst/rtsp-server/rtsp-media.c:
6197 * gst/rtsp-server/rtsp-mount-points.c:
6198 * gst/rtsp-server/rtsp-server.c:
6199 * gst/rtsp-server/rtsp-token.c:
6200 rtsp-*: Refer to NULL as a constant in comments
6202 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6204 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6206 rtsp-*: Fix type name typos in comments
6207 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
6208 * rtsp-auth: Refer to part of constant name as text
6209 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
6210 * rtsp-session-media: Fix GstRTSPSessionMedia typo
6211 * rtsp-stream: Fix typo when refering to GstBin
6212 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6214 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6217 * docs/libs/gst-rtsp-server-docs.sgml:
6218 * docs/libs/gst-rtsp-server-sections.txt:
6219 docs: Improve documentation
6220 * Include annotation-glossary to quiet gtk-doc
6221 * Rename remaining ClientState -> Context
6222 * Rename object hierarchy file
6223 * Remove stale chapter references
6224 * Add missing function and object references
6225 * Include missing GstRTSPAddressPoolResult
6226 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6228 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
6230 * gst/rtsp-server/rtsp-client.c:
6231 * gst/rtsp-server/rtsp-server.c:
6232 * gst/rtsp-server/rtsp-session-pool.c:
6233 * gst/rtsp-server/rtsp-session.c:
6234 * gst/rtsp-server/rtsp-stream.c:
6235 rtsp-server: sprinkle some allow-none annotations for g-i
6237 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
6239 * gst/rtsp-server/rtsp-stream.c:
6240 * gst/rtsp-server/rtsp-stream.h:
6241 stream: add method to filter transports
6242 Add a method to safely iterate and collect the stream transports
6243 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
6245 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
6247 * gst/rtsp-server/rtsp-client.c:
6248 * gst/rtsp-server/rtsp-server.c:
6249 * gst/rtsp-server/rtsp-session-pool.c:
6250 * gst/rtsp-server/rtsp-session.c:
6251 rtsp: allow NULL func in filters
6252 Passing a null function make the filters return a list of
6255 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
6257 * gst/rtsp-server/rtsp-address-pool.c:
6258 * tests/check/gst/addresspool.c:
6259 address-pool: fix address increment
6260 Use a guint instead of guint8 to increment the address. It's still not
6261 completely correct because a guint might not be able to hold the complete
6262 address range, but that's an enhacement for later.
6263 Add unit test to test improved behaviour.
6264 https://bugzilla.gnome.org/show_bug.cgi?id=708237
6266 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
6268 * gst/rtsp-server/rtsp-client.c:
6269 * tests/check/gst/client.c:
6270 client: allow absolute path in requests
6271 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
6273 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
6275 * gst/rtsp-server/rtsp-client.c:
6276 * gst/rtsp-server/rtsp-client.h:
6277 client: make make_path_from_uri a vmethod
6279 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6281 * docs/libs/gst-rtsp-server-sections.txt:
6282 * gst/rtsp-server/rtsp-stream.c:
6283 * gst/rtsp-server/rtsp-stream.h:
6284 * tests/check/Makefile.am:
6285 * tests/check/gst/stream.c:
6286 stream: Add functions to get rtp and rtcp sockets
6287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
6289 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6291 * gst/rtsp-server/rtsp-context.c:
6292 * gst/rtsp-server/rtsp-context.h:
6293 context: defing a GType for the context
6294 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
6296 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6298 * gst/rtsp-server/Makefile.am:
6299 * gst/rtsp-server/rtsp-auth.c:
6300 * gst/rtsp-server/rtsp-context.c:
6301 * gst/rtsp-server/rtsp-media.c:
6302 * gst/rtsp-server/rtsp-mount-points.c:
6303 * gst/rtsp-server/rtsp-server.h:
6304 * gst/rtsp-server/rtsp-session-media.c:
6305 * gst/rtsp-server/rtsp-session.c:
6306 * gst/rtsp-server/rtsp-stream.c:
6307 Fixed several GIR warnings
6309 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
6311 * gst/rtsp-server/rtsp-auth.c:
6314 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6316 * tests/check/Makefile.am:
6317 * tests/check/gst/token.c:
6318 tests: Add unit tests for token
6319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6321 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6323 * gst/rtsp-server/rtsp-token.c:
6324 token: Validate args for gst_rtsp_token_is_allowed
6325 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
6327 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6329 * gst/rtsp-server/rtsp-token.c:
6330 token: Fix bug when creating empty token
6331 We always want to have a valid GstStructure in the token.
6332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6334 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6336 * gst/rtsp-server/rtsp-thread-pool.c:
6337 thread-pool: avoid race in shutdown
6338 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
6339 don't actually stop the mainloop ever. Solve this race by adding an idle source
6340 to the mainloop that calls the _quit. This way we immediately exit the mainloop
6341 if quit was called before we started it.
6343 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6345 * tests/check/Makefile.am:
6346 * tests/check/gst/permissions.c:
6347 tests: Add unit tests for permissions
6348 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
6350 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6352 * tests/check/gst/mediafactory.c:
6353 tests: Test mediafactory permissions
6354 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6356 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6358 * gst/rtsp-server/rtsp-permissions.c:
6359 permissions: Fix refcounting when adding/removing roles
6360 Previously a role that was removed was unreffed twice, and when
6361 replacing an existing role the replaced role was freed while still being
6362 referenced. Both bugs are now fixed.
6363 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6365 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6367 * tests/check/gst/media.c:
6368 * tests/check/gst/mediafactory.c:
6369 * tests/check/gst/rtspserver.c:
6370 tests: Check gst_rtsp_url_parse return value
6371 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6373 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
6376 Automatic update of common submodule
6377 From 865aa20 to dbedaa0
6379 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
6381 * gst/rtsp-server/rtsp-server.c:
6382 rtsp-server: Fix socket leak
6383 https://bugzilla.gnome.org/show_bug.cgi?id=710088
6385 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
6387 * gst/rtsp-server/rtsp-session-pool.c:
6388 rtsp-session-pool: Make sure session IDs are properly URI-escaped
6389 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6391 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6393 * examples/.gitignore:
6394 * examples/test-video.c:
6395 examples: fix compilation when WITH_AUTH is defined
6396 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6398 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
6401 gitignore: Add new test binary
6403 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
6405 * tests/check/Makefile.am:
6406 * tests/check/gst/threadpool.c:
6407 thread-pool: Add unit test for the thread pools
6408 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6410 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6412 * gst/rtsp-server/rtsp-thread-pool.c:
6413 thread-pool: Fix thread leak when reusing threads
6414 https://bugzilla.gnome.org/show_bug.cgi?id=709730
6416 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
6418 * gst/rtsp-server/rtsp-server.c:
6419 * tests/check/gst/rtspserver.c:
6420 tests: fixed racy behavior in rtspserver tests
6421 https://bugzilla.gnome.org/show_bug.cgi?id=710078
6423 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6425 * tests/check/gst/addresspool.c:
6426 tests: Improve address pool unit tests
6427 Add a range with mixed IPV4 and IPV6 addresses to pool.
6428 Get an IPV4 address from an IPV6-only pool.
6429 Get an IPV6 address from an IPV4-only pool.
6430 Reserve a IPV6 address from an IPV4-only pool.
6431 Check for unicast addresses in multicast-only pool.
6432 Check for unicast addresses in uni-/multicast-mixed pool.
6433 https://bugzilla.gnome.org/show_bug.cgi?id=710128
6435 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6437 * gst/rtsp-server/rtsp-client.c:
6438 client: append query string in PAUSE/PLAY/TEARDOWN as well
6440 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
6442 * gst/rtsp-server/rtsp-client.c:
6443 client: Add query to control path
6444 If the SETUP url contains a query it must be appended to the control
6445 path so that it matches any already created stream in the media. The
6446 query will also be appended to the session media path.
6448 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6450 * gst/rtsp-server/rtsp-media.c:
6451 rtsp-media: remove old line
6453 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
6455 * gst/rtsp-server/rtsp-stream.c:
6456 stream: Correct control comparison
6457 https://bugzilla.gnome.org/show_bug.cgi?id=709176
6459 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6461 * gst/rtsp-server/rtsp-media.c:
6462 media: Check dynamically if the pipeline supports seeking
6463 We should not depend on whether or not the pipeline state change
6464 returned NO_PREROLL or not. A media could dynamically change its
6465 element and switch from seekable to non seekable so it's best to test
6466 the seekable nature of the pipeline dynamically when we try to do a seek.
6468 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6470 * gst/rtsp-server/rtsp-media.c:
6471 media: Return FALSE if seeking is not supported
6473 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6475 * gst/rtsp-server/rtsp-media.c:
6476 rtsp-media: don't seek accurate by default
6477 Accurate seeking is perhaps a little overkill in the most common situation and
6478 causes some formats (mp3) over slow media to seek extremely slowly.
6480 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
6482 * tests/check/gst/rtspserver.c:
6483 tests: fix unit test
6484 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
6486 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
6488 * gst/rtsp-server/rtsp-client.c:
6489 client: Reply 400 if media cannot be constructed
6490 Reply 400 Bad Request instead of 503 Service Unavailable if media
6491 cannot be constructed in SETUP.
6492 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
6494 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
6496 * gst/rtsp-server/rtsp-client.c:
6497 client: Send setup reply once only
6498 If find_media() failed in handle_setup_request() two replies was sent.
6499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
6501 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
6504 Automatic update of common submodule
6505 From 6b03ba7 to 865aa20
6507 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
6509 * gst/rtsp-server/rtsp-server.c:
6510 server: Emit client-connected signal earlier
6511 Emit client-connected before the client ref is given to a GSource,
6512 otherwise client-connected can be emitted after the client object has
6515 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
6517 * gst/rtsp-server/rtsp-address-pool.c:
6518 * gst/rtsp-server/rtsp-address-pool.h:
6519 * gst/rtsp-server/rtsp-stream.c:
6520 * tests/check/gst/addresspool.c:
6521 addresspool: return reason of failure
6522 Let gst_rtsp_address_pool_reserve_address() return the reason why
6523 the address could not be reserved.
6524 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
6526 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
6529 autogen.sh: Sync behaviour with other GStreamer modules
6530 Allows building from outside of tree amongst other things
6532 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
6535 Automatic update of common submodule
6536 From b613661 to 6b03ba7
6538 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
6541 Automatic update of common submodule
6542 From 74a6857 to b613661
6544 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
6547 Automatic update of common submodule
6548 From 01a7a46 to 74a6857
6550 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
6552 * gst/rtsp-server/rtsp-client.c:
6553 client: Do not read beyond end of path string
6554 If the setup was done without a control url, make sure we don't try to read the
6555 non-existing control string and crash.
6557 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6559 * gst/rtsp-server/rtsp-client.c:
6560 client: Fix RTPInfo header
6561 Refactor the method to make the content_base.
6562 Use the content-base and the control url to construct the RTPInfo
6565 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6567 * gst/rtsp-server/rtsp-client.c:
6568 client: map url to path only in describe
6569 Only map the request url to a path in the DESCRIBE method. The SDP then
6570 contains the base and control urls that should be used to SETUP/PAUSE/
6571 PLAY/TEARDOWN the media.
6573 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6575 * gst/rtsp-server/rtsp-client.c:
6576 Revert "client: map URL to path in requests"
6577 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
6578 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
6579 contains the base and control urls which are used in the SETUP, PLAY,
6580 PAUSE and TEARDOWN requests.
6582 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6584 * gst/rtsp-server/rtsp-client.c:
6585 client: map URL to path in requests
6587 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6589 * gst/rtsp-server/rtsp-client.c:
6590 * gst/rtsp-server/rtsp-mount-points.c:
6591 * gst/rtsp-server/rtsp-mount-points.h:
6592 mount-points: make vmethod to make path from uri
6593 Make a vmethod to transform an url into a path. The path is then used to lookup
6594 the factory. This makes it possible to also use other bits of the url, such as
6595 the query parameters, to locate the factory.
6597 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
6599 * gst/rtsp-server/rtsp-thread-pool.c:
6600 * gst/rtsp-server/rtsp-thread-pool.h:
6601 thread-pool: Add cleanup to wait for the threadpool to finish
6602 Also fix race condition if two threads are asking for the first
6603 thread from the thread pool at once. This would case two internal
6604 GThreadPools to be created.
6605 https://bugzilla.gnome.org/show_bug.cgi?id=707753
6607 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
6609 * gst/rtsp-server/rtsp-client.c:
6610 * tests/check/gst/client.c:
6611 client: free threadpool
6612 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6614 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
6616 * tests/check/gst/mountpoints.c:
6617 mountpoints tests: unref matched factories
6618 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6620 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
6622 * tests/check/gst/media.c:
6623 media tests: unref thread pool and caps
6624 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6626 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
6628 * gst/rtsp-server/rtsp-auth.c:
6629 * gst/rtsp-server/rtsp-media-factory.c:
6630 * gst/rtsp-server/rtsp-media.c:
6631 auth, media, media-factory: unref permissions
6632 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6634 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6636 * examples/Makefile.am:
6637 Makefile: add rule for appsrc example
6639 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6641 * examples/test-appsrc.c:
6642 tests: add appsrc example
6643 Add an example on how to use appsrc to feed the server pipeline with data.
6645 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
6647 * gst/rtsp-server/rtsp-client.c:
6648 rtsp-client: remove query part from content-base string
6649 Make sure that after the control url has been resolved, it's
6650 not a part of the query-string.
6651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
6653 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6655 * gst/rtsp-server/rtsp-client.c:
6656 client: don't check url in response
6657 There is no url or method in the response to check
6659 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6661 * gst/rtsp-server/rtsp-client.c:
6662 * gst/rtsp-server/rtsp-client.h:
6663 Add handle-response signal for when we receive a GET_PARAMETER response
6665 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6667 * gst/rtsp-server/rtsp-server.c:
6668 Fix gst_rtsp_server_client_filter, using wrong variable type
6670 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
6672 * gst/rtsp-server/rtsp-media-factory-uri.c:
6673 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
6674 For AAC we need to check for framed=true instead of parsed=true.
6675 https://bugzilla.gnome.org/show_bug.cgi?id=701384
6677 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6679 * gst/rtsp-server/rtsp-stream.c:
6680 stream: optimize pipeline for protocols
6681 When TCP is not an allowed protocol for the stream, avoid creating the
6682 appsrc/appsink/queue and tee elements.
6684 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6686 * gst/rtsp-server/rtsp-media.c:
6687 media: set protocols on streams
6689 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6691 * gst/rtsp-server/rtsp-client.c:
6692 client: use protocols supported by stream
6694 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6696 * gst/rtsp-server/rtsp-media-factory.c:
6697 * gst/rtsp-server/rtsp-media.c:
6698 * gst/rtsp-server/rtsp-stream.c:
6699 media-factory: allow all protocols
6701 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6703 * gst/rtsp-server/rtsp-media.c:
6704 media: configure protocols in new streams
6706 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6708 * gst/rtsp-server/rtsp-stream.c:
6709 * gst/rtsp-server/rtsp-stream.h:
6710 stream: add protocols property
6712 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6714 * gst/rtsp-server/rtsp-media.c:
6715 rtsp-media: send state in "new-state" signal
6716 https://bugzilla.gnome.org/show_bug.cgi?id=705110
6718 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
6721 build: add subdir-objects to AM_INIT_AUTOMAKE
6722 Fixes warnings with automake 1.14
6723 https://bugzilla.gnome.org/show_bug.cgi?id=705350
6725 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6727 * docs/libs/gst-rtsp-server-sections.txt:
6728 * gst/rtsp-server/rtsp-client.c:
6729 * gst/rtsp-server/rtsp-server.c:
6730 * gst/rtsp-server/rtsp-server.h:
6731 server: add method to iterate clients of server
6733 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6735 * gst/rtsp-server/rtsp-media.c:
6736 * gst/rtsp-server/rtsp-media.h:
6737 Add vmethod for rtsp-media subclass to access rtpbin
6739 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6741 * gst/rtsp-server/rtsp-client.h:
6742 small documentation fix
6744 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6746 * gst/rtsp-server/rtsp-client.c:
6747 Do not take range header if range is invalid
6749 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6751 * docs/libs/gst-rtsp-server-sections.txt:
6752 * gst/rtsp-server/rtsp-media.c:
6753 media: add docs for new method
6755 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6757 * gst/rtsp-server/rtsp-media.c:
6758 * gst/rtsp-server/rtsp-media.h:
6759 Add API to rtsp-media set the pipeline's state
6761 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6763 * gst/rtsp-server/rtsp-media.c:
6764 Update current position/duration when gst_rtsp_media_get_range_string is called
6766 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6768 * examples/test-cgroups.c:
6769 tests: add some more docs
6771 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6773 * examples/test-cgroups.c:
6774 * gst/rtsp-server/Makefile.am:
6775 * gst/rtsp-server/rtsp-auth.c:
6776 * gst/rtsp-server/rtsp-auth.h:
6777 * gst/rtsp-server/rtsp-client.c:
6778 * gst/rtsp-server/rtsp-client.h:
6779 * gst/rtsp-server/rtsp-context.c:
6780 * gst/rtsp-server/rtsp-context.h:
6781 * gst/rtsp-server/rtsp-params.c:
6782 * gst/rtsp-server/rtsp-params.h:
6783 * gst/rtsp-server/rtsp-server.c:
6784 * gst/rtsp-server/rtsp-thread-pool.c:
6785 * gst/rtsp-server/rtsp-thread-pool.h:
6786 * tests/check/gst/client.c:
6787 ClientState -> Context
6788 Rename the clientstate to context and put the code in a separate file.
6790 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6792 * examples/test-auth.c:
6793 * gst/rtsp-server/rtsp-auth.c:
6794 * gst/rtsp-server/rtsp-auth.h:
6795 auth: add support for default token
6796 The default token is used when the user is not authenticated and can be used to
6797 give minimal permissions.
6799 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6801 * examples/test-auth.c:
6802 * gst/rtsp-server/rtsp-auth.c:
6803 auth: use defines when possible
6805 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6807 * gst/rtsp-server/rtsp-address-pool.c:
6808 address-pool: improve docs
6810 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6812 * gst/rtsp-server/rtsp-permissions.c:
6813 permissions: add the role to the copy
6815 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
6817 * gst/rtsp-server/rtsp-permissions.c:
6818 permissions: Also copy the roles
6820 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
6822 * gst/rtsp-server/rtsp-permissions.c:
6823 permissions: Make it build
6825 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6827 * gst/rtsp-server/rtsp-address-pool.h:
6830 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6832 * docs/libs/gst-rtsp-server-sections.txt:
6833 * gst/rtsp-server/rtsp-auth.c:
6834 * gst/rtsp-server/rtsp-auth.h:
6835 * gst/rtsp-server/rtsp-media.c:
6836 * gst/rtsp-server/rtsp-session-media.c:
6837 * gst/rtsp-server/rtsp-stream-transport.c:
6838 * gst/rtsp-server/rtsp-stream-transport.h:
6839 * gst/rtsp-server/rtsp-stream.c:
6840 * tests/check/gst/client.c:
6843 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6845 * docs/libs/gst-rtsp-server-sections.txt:
6846 * gst/rtsp-server/rtsp-address-pool.c:
6847 * gst/rtsp-server/rtsp-address-pool.h:
6848 * tests/check/gst/addresspool.c:
6849 * tests/check/gst/rtspserver.c:
6850 address-pool: cleanups
6851 Remove redundant method, improve docs.
6853 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * docs/libs/gst-rtsp-server-sections.txt:
6856 * gst/rtsp-server/rtsp-auth.h:
6857 * gst/rtsp-server/rtsp-permissions.c:
6858 * gst/rtsp-server/rtsp-permissions.h:
6859 * gst/rtsp-server/rtsp-token.c:
6862 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6864 * gst/rtsp-server/rtsp-permissions.c:
6865 permissions: implement _remove_role
6867 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6869 * gst/rtsp-server/rtsp-permissions.c:
6870 permissions: update docs
6872 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6874 * tests/check/gst/client.c:
6875 tests: simplify tests
6876 Client settings are now disabled by default so we don't need an auth
6877 module to disable them.
6879 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6881 * gst/rtsp-server/rtsp-auth.c:
6882 auth: add default authorizations
6883 When no auth module is specified, use our table of defaults to look up the
6884 default value of the check instead of always allowing everything. This was
6885 we can disallow client settings by default.
6887 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6890 README: update readme
6892 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6894 * gst/rtsp-server/rtsp-thread-pool.c:
6895 * gst/rtsp-server/rtsp-thread-pool.h:
6896 thread-pool: add more docs
6898 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6900 * gst/rtsp-server/rtsp-thread-pool.c:
6901 * gst/rtsp-server/rtsp-thread-pool.h:
6902 thread-pool: fix race in thread reuse
6903 If we try to reuse a thread right after we made it stop, we end up using a
6904 stopped thread. Catch this case and only reuse threads that are not stopping.
6906 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6908 * gst/rtsp-server/rtsp-server.c:
6909 server: add small debug
6911 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6913 * tests/check/gst/client.c:
6915 Add some permissions to media so we can use the auth and enable
6918 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6920 * gst/rtsp-server/rtsp-client.c:
6921 client: support pushed context in handle_request
6922 If we already have a pushed state, reuse it and add our own things. This makes
6923 it easier to write tests.
6925 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6927 * gst/rtsp-server/rtsp-auth.c:
6928 auth: don't auth on methods
6929 Don't authorize on methods anymore but on the resources that we
6930 try to access, this is more flexible.
6931 Move the authorization checks to where they are needed and let the
6932 check return the response on error.
6934 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6936 * gst/rtsp-server/rtsp-mount-points.c:
6937 mount-points: add some debug
6939 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6941 * tests/check/gst/client.c:
6942 tests: almost fix test
6944 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6946 * gst/rtsp-server/rtsp-auth.c:
6947 * gst/rtsp-server/rtsp-auth.h:
6948 * gst/rtsp-server/rtsp-client.c:
6949 * gst/rtsp-server/rtsp-client.h:
6950 * gst/rtsp-server/rtsp-server.c:
6951 * gst/rtsp-server/rtsp-server.h:
6952 auth: let the auth module check client_settings
6953 Let the auth module decide if client settings are allowed for the
6956 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6958 * gst/rtsp-server/rtsp-token.c:
6959 * gst/rtsp-server/rtsp-token.h:
6960 token: add method to check boolean permission
6962 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6964 * examples/test-auth.c:
6965 * examples/test-cgroups.c:
6966 * gst/rtsp-server/rtsp-token.c:
6967 * gst/rtsp-server/rtsp-token.h:
6968 token: simplify token constructor
6969 Use variable arguments to make easier API.
6971 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6973 * examples/test-auth.c:
6974 * examples/test-cgroups.c:
6975 * gst/rtsp-server/rtsp-media-factory.c:
6976 * gst/rtsp-server/rtsp-media-factory.h:
6977 media-factory: add convenience API for factory
6979 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6981 * examples/test-auth.c:
6982 * examples/test-cgroups.c:
6983 * gst/rtsp-server/rtsp-permissions.c:
6984 * gst/rtsp-server/rtsp-permissions.h:
6985 permissions: simplify API a little
6986 Avoid passing GstStructure in the add_role method, use varargs instead
6987 to construct the structure behind the scenes. We can then also use the
6988 structure name as the role and simplify some more logic.
6990 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6992 * gst/rtsp-server/rtsp-auth.c:
6995 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6997 * gst/rtsp-server/rtsp-auth.c:
6998 * gst/rtsp-server/rtsp-auth.h:
6999 * gst/rtsp-server/rtsp-client.c:
7000 auth: handle unauthorized response
7001 Move handling of the unauthorized response to the auth module, it can add
7002 the appropriate headers to request authorization for the required method
7003 much better than the client.
7005 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7007 * gst/rtsp-server/rtsp-client.c:
7008 * gst/rtsp-server/rtsp-client.h:
7009 client: allow for sending any message, not only requests
7010 Change the _send_request() method to _send_message() so that we
7011 can both send requests and replies.
7013 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7015 * docs/libs/gst-rtsp-server-sections.txt:
7016 * gst/rtsp-server/rtsp-server.h:
7019 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7021 * examples/test-video.c:
7022 * gst/rtsp-server/rtsp-auth.c:
7023 * gst/rtsp-server/rtsp-auth.h:
7024 * gst/rtsp-server/rtsp-server.c:
7025 * gst/rtsp-server/rtsp-server.h:
7026 auth: move TLS handling to auth module
7027 Remove the TLS settings on the server and move it to the auth module because
7028 that is where security related bits go.
7030 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7032 * gst/rtsp-server/rtsp-client.c:
7033 * gst/rtsp-server/rtsp-client.h:
7034 client: add state push/pop
7036 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7038 * gst/rtsp-server/rtsp-client.c:
7039 * gst/rtsp-server/rtsp-client.h:
7040 client: add connection to state
7042 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7044 * gst/rtsp-server/rtsp-mount-points.c:
7045 mount-points: fix debug
7047 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7049 * tests/check/gst/media.c:
7050 tests: fix media test
7052 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7054 * gst/rtsp-server/rtsp-thread-pool.c:
7055 thread-pool: we don't require a state
7057 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7059 * gst/rtsp-server/rtsp-server.c:
7060 server: let context ref the server
7061 So that we don't risk losing the server object early anc crash.
7063 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7065 * tests/check/gst/client.c:
7066 tests: fix client test
7068 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7071 * docs/libs/gst-rtsp-server-docs.sgml:
7072 * docs/libs/gst-rtsp-server-sections.txt:
7073 * gst/rtsp-server/rtsp-address-pool.c:
7074 * gst/rtsp-server/rtsp-auth.c:
7075 * gst/rtsp-server/rtsp-client.c:
7076 * gst/rtsp-server/rtsp-client.h:
7077 * gst/rtsp-server/rtsp-media-factory-uri.c:
7078 * gst/rtsp-server/rtsp-media-factory.c:
7079 * gst/rtsp-server/rtsp-media-factory.h:
7080 * gst/rtsp-server/rtsp-media.c:
7081 * gst/rtsp-server/rtsp-mount-points.c:
7082 * gst/rtsp-server/rtsp-params.c:
7083 * gst/rtsp-server/rtsp-permissions.c:
7084 * gst/rtsp-server/rtsp-sdp.c:
7085 * gst/rtsp-server/rtsp-server.c:
7086 * gst/rtsp-server/rtsp-server.h:
7087 * gst/rtsp-server/rtsp-session-media.c:
7088 * gst/rtsp-server/rtsp-session-pool.c:
7089 * gst/rtsp-server/rtsp-session.c:
7090 * gst/rtsp-server/rtsp-stream-transport.c:
7091 * gst/rtsp-server/rtsp-stream.c:
7092 * gst/rtsp-server/rtsp-thread-pool.c:
7093 * gst/rtsp-server/rtsp-token.c:
7096 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7098 * gst/rtsp-server/rtsp-session-pool.c:
7099 * gst/rtsp-server/rtsp-session-pool.h:
7100 session-pool: make vmethod to create a session
7101 Make a vmethod to create a sessions so that subclasses can create
7102 custom session objects
7104 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7106 * gst/rtsp-server/rtsp-auth.c:
7107 * gst/rtsp-server/rtsp-media-factory.h:
7108 * gst/rtsp-server/rtsp-media.h:
7109 * gst/rtsp-server/rtsp-mount-points.h:
7110 * gst/rtsp-server/rtsp-session-pool.h:
7111 * gst/rtsp-server/rtsp-stream.h:
7114 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7116 * docs/libs/gst-rtsp-server-docs.sgml:
7117 * docs/libs/gst-rtsp-server-sections.txt:
7118 * gst/rtsp-server/rtsp-address-pool.c:
7119 * gst/rtsp-server/rtsp-address-pool.h:
7120 * gst/rtsp-server/rtsp-auth.c:
7121 * gst/rtsp-server/rtsp-client.h:
7122 * gst/rtsp-server/rtsp-media-factory.h:
7123 * gst/rtsp-server/rtsp-media.c:
7124 * gst/rtsp-server/rtsp-media.h:
7125 * gst/rtsp-server/rtsp-permissions.c:
7126 * gst/rtsp-server/rtsp-permissions.h:
7127 * gst/rtsp-server/rtsp-server.h:
7128 * gst/rtsp-server/rtsp-session-media.c:
7129 * gst/rtsp-server/rtsp-session-media.h:
7130 * gst/rtsp-server/rtsp-session-pool.h:
7131 * gst/rtsp-server/rtsp-session.h:
7132 * gst/rtsp-server/rtsp-stream-transport.h:
7133 * gst/rtsp-server/rtsp-stream.c:
7134 * gst/rtsp-server/rtsp-thread-pool.h:
7137 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7140 * examples/Makefile.am:
7141 configure: compile cgroup example conditionally
7142 Only compile the cgroup example when we have libcgroup
7144 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7147 * examples/Makefile.am:
7148 * examples/test-cgroups.c:
7149 examples: add cgroups example
7151 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7153 * tests/check/gst/rtspserver.c:
7154 tests: fix compilation
7156 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7158 * gst/rtsp-server/rtsp-thread-pool.c:
7159 thread-pool: fix vmethod invocation
7161 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7163 * gst/rtsp-server/rtsp-thread-pool.c:
7164 * gst/rtsp-server/rtsp-thread-pool.h:
7165 thread-pool: store thread type in thread
7167 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7169 * gst/rtsp-server/rtsp-client.c:
7170 client: pass thread from pool to media _prepare
7171 Get a thread from the configured threadpool and pass it to the prepare method of
7174 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-media.c:
7177 * gst/rtsp-server/rtsp-media.h:
7178 media: Accept a thread in _prepare
7179 Remove out own threadpool handling and use the provided thread and
7180 maincontext for the bus messages and the state changes.
7182 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7184 * gst/rtsp-server/rtsp-server.c:
7185 server: configure client thread pool
7187 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7189 * gst/rtsp-server/rtsp-client.c:
7190 * gst/rtsp-server/rtsp-client.h:
7191 client: add method to configure thread pool
7193 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7195 * gst/rtsp-server/rtsp-client.h:
7196 * gst/rtsp-server/rtsp-server.c:
7197 * gst/rtsp-server/rtsp-server.h:
7198 server: use thread pool
7199 Use the thread pool instead of doing our own thing.
7201 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * gst/rtsp-server/Makefile.am:
7204 * gst/rtsp-server/rtsp-thread-pool.c:
7205 * gst/rtsp-server/rtsp-thread-pool.h:
7206 thread-pool: add object to manage threads
7207 Add an object to manage the client and media threads.
7209 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7211 * gst/rtsp-server/rtsp-auth.c:
7212 auth: debug authorization check
7214 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7216 * gst/rtsp-server/rtsp-media.c:
7217 media: start media pipeline in context
7218 Start the media pipeline in the provided context (or our default one
7219 when NULL). This makes sure that we run the bus thread in this context and that
7220 all media threads are children of this context.
7222 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * gst/rtsp-server/rtsp-media-factory.c:
7225 factory: pass permissions to media by default
7227 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7229 * examples/test-auth.c:
7230 test: add permissions to auth test
7231 Ass some permissions to the media factory in the test.
7233 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7235 * gst/rtsp-server/rtsp-auth.c:
7236 * gst/rtsp-server/rtsp-auth.h:
7237 * gst/rtsp-server/rtsp-client.c:
7238 auth: simplify auth checks
7239 Remove client from methods, it's now in the state
7240 Perform the check specified by the string, use the information from the
7241 thread local context.
7243 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7245 * gst/rtsp-server/rtsp-client.c:
7246 * gst/rtsp-server/rtsp-client.h:
7247 client: add state to current thread
7248 Add the client to the ClientState object.
7249 Place the ClientState on the current thread.
7251 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-media-factory.c:
7254 * gst/rtsp-server/rtsp-media-factory.h:
7255 * gst/rtsp-server/rtsp-media.c:
7256 * gst/rtsp-server/rtsp-media.h:
7257 media: make it possible to set permissions
7258 Make it possible to set permissions on media and media factory objects
7260 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7262 * gst/rtsp-server/Makefile.am:
7263 * gst/rtsp-server/rtsp-permissions.c:
7264 * gst/rtsp-server/rtsp-permissions.h:
7265 permissions: add permissions object
7266 Add a mini object to store permissions based on a role.
7268 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7270 * examples/test-auth.c:
7271 * gst/rtsp-server/rtsp-auth.c:
7272 * gst/rtsp-server/rtsp-auth.h:
7273 * gst/rtsp-server/rtsp-client.c:
7274 auth: add auth checks
7275 Add an enum with auth checks and implement the checks in the auth object.
7276 Perform the checks from the client.
7278 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7280 * examples/test-auth.c:
7281 * gst/rtsp-server/rtsp-auth.c:
7282 * gst/rtsp-server/rtsp-auth.h:
7283 * gst/rtsp-server/rtsp-client.h:
7284 auth: use the token after authentication
7285 After we authenticated a user, keep the Token around in the state.
7287 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7289 * gst/rtsp-server/rtsp-client.c:
7290 * gst/rtsp-server/rtsp-media.c:
7291 * gst/rtsp-server/rtsp-media.h:
7292 * tests/check/gst/media.c:
7293 media: add optional context for bus messages
7294 Add an optional mainloop to _prepare that will handle the bus messages instead
7295 of always using the shared mainloop.
7297 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7299 * gst/rtsp-server/Makefile.am:
7300 * gst/rtsp-server/rtsp-token.c:
7301 * gst/rtsp-server/rtsp-token.h:
7302 token: add authorization token
7303 Add a simply miniobject that contains the authorizations. The object contains a
7304 GstStructure that hold all authorization fields. When a user is authenticated,
7305 the auth module will create a Token for the user. The token is then used to
7306 check what operations the user is allowed to do and various other configuration
7309 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7311 * examples/test-auth.c:
7312 * gst/rtsp-server/rtsp-auth.c:
7313 * gst/rtsp-server/rtsp-auth.h:
7314 * gst/rtsp-server/rtsp-client.c:
7315 * gst/rtsp-server/rtsp-client.h:
7316 * gst/rtsp-server/rtsp-media-factory.c:
7317 * gst/rtsp-server/rtsp-media-factory.h:
7318 * gst/rtsp-server/rtsp-media.c:
7319 * gst/rtsp-server/rtsp-media.h:
7320 auth: remove auth from media and factory
7321 Remove the auth object from media and factory. We want to have the RTSPClient
7322 authenticate and authorize resources, there is no need to place another auth
7323 manager on the media/factory.
7325 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7327 * examples/test-auth.c:
7328 * gst/rtsp-server/rtsp-auth.c:
7329 * gst/rtsp-server/rtsp-auth.h:
7330 * gst/rtsp-server/rtsp-client.h:
7331 auth: add support for multiple basic auth tokens
7332 Make it possible to add multiple basic authorisation tokens to one authorization
7333 object. Associate with each token an authorization group that will define what
7334 capabilities are allowed.
7336 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7338 * gst/rtsp-server/rtsp-client.c:
7339 client: error out on non-aggregate control
7340 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
7342 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7344 * gst/rtsp-server/rtsp-client.c:
7345 client: rework setup request a little
7346 Cache the media in DESCRIBE based on the longest matching path with the uri
7347 that we can find in the mount points.
7348 Rework the setup request a little to get the media from the session or from
7349 the longest matching path, this way we can derive the control string as
7350 everything after the path instead of hardcoding it.
7351 Find the stream based on the control string and only open a session when all
7354 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7356 * gst/rtsp-server/rtsp-media.c:
7357 * gst/rtsp-server/rtsp-media.h:
7358 media: add method to find a stream by control url
7360 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7362 * gst/rtsp-server/rtsp-stream.c:
7363 * gst/rtsp-server/rtsp-stream.h:
7364 stream: add method to check control url of stream
7366 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7368 * gst/rtsp-server/rtsp-client.c:
7369 * gst/rtsp-server/rtsp-session-media.c:
7370 * gst/rtsp-server/rtsp-session-media.h:
7371 * gst/rtsp-server/rtsp-session.c:
7372 * gst/rtsp-server/rtsp-session.h:
7373 session: use path matching for session media
7374 Use a path string instead of a uri to lookup session media in the sessions. Also
7375 use path matching to find the largest possible path that matches.
7377 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7379 * gst/rtsp-server/rtsp-client.c:
7380 * gst/rtsp-server/rtsp-mount-points.c:
7381 * gst/rtsp-server/rtsp-mount-points.h:
7382 * tests/check/gst/mountpoints.c:
7383 mount-points: remove useless vmethod
7384 Making lookups in the mount points should not be done with a URL, if there is a
7385 mapping to be done from URL to mount points, we'll need to do it somewhere
7388 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7390 * gst/rtsp-server/rtsp-mount-points.c:
7391 * gst/rtsp-server/rtsp-mount-points.h:
7392 * tests/check/gst/mountpoints.c:
7393 mount-points: improve mount point searching
7394 Use a GSequence to keep track of the mount points.
7395 Match a URL to the longest matching registered mount point. This should be the
7396 URL to perform aggreagate control and the remainder is the stream specific
7398 Add some unit tests for this.
7400 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
7402 * gst/rtsp-server/Makefile.am:
7403 rtsp-server: Allow building of static library
7405 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7407 * tests/check/gst/mediafactory.c:
7408 tests: fix compilation
7410 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7412 * gst/rtsp-server/rtsp-sdp.c:
7413 sdp: get control string from stream
7414 Use the control string as configured in the stream.
7416 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7418 * gst/rtsp-server/rtsp-stream.c:
7419 * gst/rtsp-server/rtsp-stream.h:
7420 stream: add methods and property to set control string
7422 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7424 * gst/rtsp-server/rtsp-client.c:
7426 Rename variables for clarity
7427 Keep media in state when we can
7429 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/rtsp-server/rtsp-client.c:
7432 * gst/rtsp-server/rtsp-stream.c:
7433 * gst/rtsp-server/rtsp-stream.h:
7434 stream: add more support for IPv6
7435 Rename _get_address to _get_multicast_address in GstRTSPStream to
7436 make it clear that this function only deals with multicast.
7437 Make it possible to have both an IPv4 and IPv6 multicast address on
7438 a stream. Give the client an IPv4 or IPv6 address depending on the
7439 address it used to connect to the server.
7440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
7442 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7444 * gst/rtsp-server/rtsp-client.c:
7447 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7449 * gst/rtsp-server/rtsp-stream.c:
7450 stream: handle failed port allocation
7451 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
7452 can't allocate any family at all. Also keep track of what port families we
7454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
7456 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7458 * gst/rtsp-server/rtsp-stream.c:
7459 stream: improve docs
7461 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * gst/rtsp-server/rtsp-stream-transport.c:
7464 stream-transport: remove old if 0 block
7466 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
7468 * tests/check/gst/client.c:
7470 gst_rtsp_client_get_uri() has been removed
7471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
7473 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7475 * gst/rtsp-server/rtsp-client.c:
7476 * gst/rtsp-server/rtsp-client.h:
7477 client: add method to filter managed sessions
7478 Add a method to filter the sessions managed by this client connection.
7479 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
7481 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7483 * gst/rtsp-server/rtsp-client.c:
7484 * gst/rtsp-server/rtsp-client.h:
7485 client: remove _get_uri() method
7486 Remove the get_uri() method on the client. A client has no uri, the uri
7487 property is an internal property to manage the last cached media for
7490 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7492 * gst/rtsp-server/rtsp-media-factory.h:
7493 media-factory: fix typo
7495 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7497 * gst/rtsp-server/rtsp-media.c:
7498 rtsp-media: Do not leak the query in default_query_stop
7499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
7501 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7503 * gst/rtsp-server/rtsp-media.c:
7504 media: don't unlock when conversion fails
7505 Don't unlock the state lock when conversion fails because it was not locked.
7507 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7509 * gst/rtsp-server/rtsp-media.c:
7510 * gst/rtsp-server/rtsp-media.h:
7511 Add query_position and query_stop vmethods to rtsp-media
7513 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7515 * gst/rtsp-server/rtsp-media.c:
7516 Fix typo in property install for rtsp-media's time-provider
7518 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7520 * gst/rtsp-server/rtsp-client.c:
7521 * gst/rtsp-server/rtsp-client.h:
7522 client: clean some variables
7523 Clean some variables and add some guards to _send_request()
7525 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7527 * gst/rtsp-server/rtsp-client.c:
7528 * gst/rtsp-server/rtsp-client.h:
7529 Add gst_rtsp_client_send_request API
7530 This makes it possible to send arbitrary messages to a client, such as
7531 SET_PARAMETER or GET_PARAMETER
7533 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7535 * gst/rtsp-server/rtsp-media.c:
7536 * gst/rtsp-server/rtsp-media.h:
7537 media: add _get_element() method
7538 Add method to get the element used when creating the media.
7539 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
7541 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7543 * gst/rtsp-server/rtsp-media.c:
7546 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7548 * gst/rtsp-server/rtsp-stream.c:
7549 * gst/rtsp-server/rtsp-stream.h:
7550 stream: allow access to the rtp session
7551 https://bugzilla.gnome.org/show_bug.cgi?id=703004
7553 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
7555 * gst/rtsp-server/rtsp-stream.c:
7556 * gst/rtsp-server/rtsp-stream.h:
7557 dscp qos support in gst-rtsp-stream
7558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
7560 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7562 * tests/check/gst/rtspserver.c:
7564 Actually do what the comment says. Also keep the old code around, not sure what
7565 should happen when you get a 454 from a TEARDOWN, does it close the connection?
7566 it currently doesn't.
7568 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7570 * gst/rtsp-server/rtsp-client.c:
7571 client: also watch newly created session
7572 When we newly created a session, start watching it immediately instead of
7573 on the next request.
7575 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
7577 * tests/check/gst/client.c:
7578 tests: add unit test for new-session
7579 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
7581 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7583 * gst/rtsp-server/rtsp-client.c:
7584 client: emit new-session when new session is created
7585 Only emit new-session when we created a new session for a client, not when a
7586 client picked up a previous session.
7587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
7589 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
7591 * gst/rtsp-server/rtsp-client.c:
7592 client: handle asterisk as path in requests
7593 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
7595 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7597 * gst/rtsp-server/rtsp-media.c:
7598 media: handle segment query format mismatch
7599 It's possible that the segment query returns with a different format than what
7600 we asked for, handle this case also.
7602 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
7604 * gst/rtsp-server/rtsp-media.c:
7605 media: use segment stop in collect_media_stats
7606 Use segment stop instead of duration as range end point.
7607 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
7609 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7611 * gst/rtsp-server/rtsp-media.c:
7612 * tests/check/gst/media.c:
7613 rtsp-media: Do not leak the element in take_pipeline
7614 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
7616 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
7618 * gst/rtsp-server/rtsp-client.c:
7619 * gst/rtsp-server/rtsp-client.h:
7620 rtsp-client: Make configure_client_transport virtual
7621 This patch makes configure_client_transport virtual. The functionality is
7622 needed to handle some weird clients sending multicast transport settings as url
7624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
7626 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7628 * gst/rtsp-server/rtsp-client.c:
7629 * gst/rtsp-server/rtsp-client.h:
7630 rtsp-client: Make param_set and param_get virtual
7631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
7633 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
7635 * gst/rtsp-server/rtsp-client.c:
7636 * gst/rtsp-server/rtsp-media.c:
7637 * gst/rtsp-server/rtsp-media.h:
7638 media: convert_range replaces get_range_times
7639 get_range_times worked for handling UTC ranges for seeks, but we also
7640 need to convert back from NPT to the requested unit in
7641 get_range_string. convert_range is now used for both.
7642 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
7644 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7646 * gst/rtsp-server/rtsp-client.c:
7647 * gst/rtsp-server/rtsp-sdp.c:
7648 * gst/rtsp-server/rtsp-sdp.h:
7649 sdp: cleanup sdp info
7650 We don't need to pass the proto, we can more easily check a boolean.
7651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
7653 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
7655 * gst/rtsp-server/rtsp-sdp.c:
7656 use 0.0.0.0 or :: for c= line instead of server address
7658 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
7660 * gst/rtsp-server/rtsp-client.c:
7661 use local address, not remote, in SDP
7662 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
7664 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7667 Automatic update of common submodule
7668 From 098c0d7 to 01a7a46
7670 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
7672 * gst/rtsp-server/rtsp-media.c:
7673 * gst/rtsp-server/rtsp-media.h:
7674 media: possibility to override range time conversion
7675 Make it possible to override the conversion from GstRTSPTimeRange to
7676 GstClockTimes, that is done before seeking on the media
7677 pipeline. Overriding can be useful for UTC ranges, where the default
7678 conversion gives nanoseconds since 1900.
7679 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
7681 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7683 * gst/rtsp-server/rtsp-server.c:
7684 * gst/rtsp-server/rtsp-server.h:
7685 rtsp-server: Expose the use_client_settings API
7686 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
7688 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
7690 * gst/rtsp-server/rtsp-client.c:
7691 * gst/rtsp-server/rtsp-stream.c:
7692 * gst/rtsp-server/rtsp-stream.h:
7693 rtspstream: handle both ipv4 and ipv6 clients
7694 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
7696 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7698 * gst/rtsp-server/rtsp-sdp.c:
7699 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
7700 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
7701 We already have a way to place extra attributes in the SDP by using a string
7702 property with prefix x- or a- in the caps.
7704 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7706 * gst/rtsp-server/rtsp-sdp.c:
7707 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
7708 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
7709 We already have a way to place extra attributes in the SDP, just make a string
7710 property in the payloader with a- or x- prefix.
7712 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7714 * gst/rtsp-server/rtsp-sdp.c:
7715 rtsp: place a- and x- properties as attributes
7716 application/x-rtp has properties with a- and x- prefixes that should be
7717 placed as attributes in the SDP for the media instead of being added to the
7720 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7722 * examples/Makefile.am:
7723 * examples/test-video.c:
7724 example: add TLS example
7726 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7728 * gst/rtsp-server/rtsp-server.c:
7729 * gst/rtsp-server/rtsp-server.h:
7730 server: add support for TLS
7731 Add methods to set and get a TLS certificate.
7732 Add vmethod to configure a new connection. By default, configure the TLS
7733 certificate in a new connection if needed.
7735 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7737 * gst/rtsp-server/rtsp-server.c:
7738 * gst/rtsp-server/rtsp-server.h:
7739 server: remove accept_client vmethod
7740 This vmethod is not very useful so remove it.
7742 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7744 * gst/rtsp-server/rtsp-server.c:
7745 server: don't crash on NULL GError
7747 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
7749 * gst/rtsp-server/rtsp-session-pool.c:
7750 rtsp-session-pool: corrected session timeout detection
7751 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
7753 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7755 * gst/rtsp-server/rtsp-client.c:
7756 client: improve debug
7758 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7760 * gst/rtsp-server/rtsp-client.c:
7761 * gst/rtsp-server/rtsp-client.h:
7762 * gst/rtsp-server/rtsp-server.c:
7763 server: refactor connection setup
7764 Let the server accept the socket connection and construct a GstRTSPConnection
7765 from it. Remove the code from the client and let the client only deal with
7766 a fully configure GstRTSPConnection object.
7767 We will need this later when the server will configure the connection for
7770 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7772 * gst/rtsp-server/rtsp-stream.c:
7773 stream: keep the transport object alive
7774 Keep the transport object alive while we have it as qdata on the
7777 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
7779 * gst/rtsp-server/rtsp-client.c:
7780 * gst/rtsp-server/rtsp-server.c:
7781 rtsp-server: Do not crash on nmapping of server
7782 * generate error when gst_rtsp_connection_accept fails
7783 * do not stop accepting incoming connections because
7784 accepting a client fails
7785 https://bugzilla.gnome.org/show_bug.cgi?id=701072
7787 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
7789 * gst/rtsp-server/rtsp-client.c:
7790 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
7791 https://bugzilla.gnome.org/show_bug.cgi?id=700953
7793 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7795 * gst/rtsp-server/rtsp-sdp.c:
7796 rtsp-sdp: Parse framerate caps field and set SDP attribute
7797 The SDP attribute and its format is described in RFC4566.
7798 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7800 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
7802 * gst/rtsp-server/rtsp-sdp.c:
7803 rtsp-sdp: Parse width/height from caps and set SDP attribute
7804 The SDP attribute and its format is described in RFC6064.
7805 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7807 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
7809 * gst/rtsp-server/rtsp-sdp.c:
7810 * tests/check/gst/client.c:
7811 rtsp-sdp: add bandwidth line
7812 https://bugzilla.gnome.org/show_bug.cgi?id=699220
7814 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7817 Automatic update of common submodule
7818 From 5edcd85 to 098c0d7
7820 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7822 * tests/check/gst/media.c:
7823 tests: add dynamic payloader prepare/unprepare check
7825 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7827 * gst/rtsp-server/rtsp-media.c:
7828 media: release lock when removing fakesink
7830 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7832 * gst/rtsp-server/rtsp-stream.c:
7833 stream: set elements to NULL before removing
7834 When removing a stream, set the elements to NULL first. This avoids
7835 element-is-not-in-NULL-state errors when we dispose the elements.
7837 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7840 Automatic update of common submodule
7841 From 3cb3d3c to 5edcd85
7843 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7845 * gst/rtsp-server/rtsp-media.c:
7846 * gst/rtsp-server/rtsp-media.h:
7847 media: listen to pad-removed signals
7848 Listen to the pad-removed signal and remove the stream associated with the
7850 Add signal to be notified of the removed pad.
7851 Remove the fakesink in unprepare()
7852 Fix signatures of the signal methods
7854 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7856 * examples/test-sdp.c:
7857 tests: add example of reusable pipelines
7859 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7861 * gst/rtsp-server/rtsp-stream.c:
7862 * gst/rtsp-server/rtsp-stream.h:
7863 stream: add method to get the srcpad
7865 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7867 * tests/check/gst/media.c:
7868 check: add media prepare/unprepare test
7869 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7871 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
7873 * gst/rtsp-server/rtsp-media.c:
7874 media: disconnect from signal handlers in unprepare()
7875 We connected to the pad-added and no-more-pads signals in prepare() so
7876 we need to disconnect from them in unprepare().
7877 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7879 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7881 * gst/rtsp-server/rtsp-media.c:
7882 media: don't free streams array
7883 Don't free the streams array in the unprepare() method, they were not
7885 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7887 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
7889 * gst/rtsp-server/rtsp-media.c:
7890 media: don't unref the pipeline in unprepare
7891 Unprepare() should undo what prepare() does. Because the pipeline is
7892 not created in prepare(), we should not unref it in unprepare()
7894 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
7896 * gst/rtsp-server/rtsp-stream.c:
7897 stream: clear session and caps for reuse
7898 Set the session and caps to NULL after unref otherwise we might unref
7900 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7902 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
7904 * gst/rtsp-server/rtsp-client.c:
7905 client: send out teardown signal before tearing down
7906 The advantage is that in the signal handler you get direct access to
7907 information about what streams are about to get torn down (in the
7908 GstRTSPClientState).
7909 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
7911 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
7913 * gst/rtsp-server/rtsp-client.c:
7914 * gst/rtsp-server/rtsp-client.h:
7915 client: expose connection
7916 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
7918 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
7921 Automatic update of common submodule
7922 From aed87ae to 3cb3d3c
7924 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7926 * gst/rtsp-server/rtsp-media.c:
7927 * gst/rtsp-server/rtsp-media.h:
7928 * gst/rtsp-server/rtsp-session-media.c:
7929 * gst/rtsp-server/rtsp-session-media.h:
7930 media: add method to get the base_time of the pipeline
7931 Together with a shared clock, this base-time could eventually be sent to
7932 the client so that it can reconstruct the exact running-time of the clock
7935 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7937 * gst/rtsp-server/Makefile.am:
7938 * gst/rtsp-server/rtsp-media.c:
7939 * gst/rtsp-server/rtsp-media.h:
7940 * gst/rtsp-server/rtsp-sdp.c:
7941 media: add GstNetTimeProvider support
7942 Add a property to let the media provide a GstNetTimeProvider for its clock.
7943 Make methods to get the clock and nettimeprovider
7944 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
7945 provider and also the current time of the clock. This should make it possible
7946 for (GStreamer) clients to slave their clock to the server clock.
7948 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
7951 Automatic update of common submodule
7952 From 04c7a1e to aed87ae
7954 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7956 * gst/rtsp-server/rtsp-media.c:
7957 media: wait for buffering to complete
7958 Wait for buffering to complete before changing the state to the target state.
7960 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7962 * gst/rtsp-server/rtsp-media.c:
7963 media: small cleanup
7965 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
7967 * tests/check/gst/rtspserver.c:
7968 tests: remove extra unref in test_setup_non_existing_stream
7969 The unref is not needed anymore, teardown runs without it.
7970 https://bugzilla.gnome.org/show_bug.cgi?id=696542
7972 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
7974 * tests/check/gst/rtspserver.c:
7975 tests: GSocketService cleanup in test_bind_already_in_use
7976 Use g_socket_service_stop so the rtspserver test stops listening for
7977 incoming connections in test_bind_already_in_use.
7978 https://bugzilla.gnome.org/show_bug.cgi?id=696541
7980 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
7982 * gst/rtsp-server/rtsp-media-factory.c:
7983 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
7984 Instead use a GWeakRef which is safe to use
7985 This is a known GLib bug, see:
7986 https://bugzilla.gnome.org/show_bug.cgi?id=667145
7988 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
7990 * gst/rtsp-server/rtsp-client.c:
7991 * gst/rtsp-server/rtsp-media.c:
7992 * gst/rtsp-server/rtsp-media.h:
7993 * gst/rtsp-server/rtsp-sdp.c:
7994 * tests/check/gst/media.c:
7995 * tests/check/gst/rtspserver.c:
7996 rtsp-media/client: Reply to PLAY request with same type of Range
7997 Remember the type of Range from the PLAY request and use the same type for
8000 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
8002 * gst/rtsp-server/rtsp-client.c:
8003 * gst/rtsp-server/rtsp-client.h:
8004 * tests/check/gst/client.c:
8005 rtsp-client: expose uri
8007 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
8009 * tests/check/gst/mediafactory.c:
8010 tests: Hold ref while creating second media
8011 To test if the media aren't shared, make sure we keep the first one while creating a second
8012 otherwise the same memory address may be reused.
8014 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
8017 configure: remove out-of-date comment
8019 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
8022 .gitignore: ignore more build files
8024 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8026 * tests/check/Makefile.am:
8027 tests: use right _LIBS variable for gst-plugins-base libs
8029 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8031 * tests/check/Makefile.am:
8032 check: add librtp to libs
8034 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
8036 * tests/check/gst/rtspserver.c:
8037 tests: Add test to check selecting a port the server will send from
8039 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
8041 * tests/check/gst/rtspserver.c:
8042 tests: Make sure packets are actually received
8044 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8046 * gst/rtsp-server/rtsp-stream.c:
8047 stream: Select unicast address from pool if appropriate
8049 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
8051 * gst/rtsp-server/rtsp-stream.c:
8052 stream: Properties are always there in Gst 1.0
8054 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8056 * tests/check/gst/addresspool.c:
8057 tests: Add tests for unicast addresses in pool
8059 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
8061 * gst/rtsp-server/rtsp-address-pool.c:
8062 * tests/check/gst/addresspool.c:
8063 address-pool: Verify that multicast addresses are used for multicast and vice-versa
8065 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
8067 * docs/libs/gst-rtsp-server-sections.txt:
8068 * gst/rtsp-server/rtsp-address-pool.c:
8069 * gst/rtsp-server/rtsp-address-pool.h:
8070 * gst/rtsp-server/rtsp-stream.c:
8071 * tests/check/gst/addresspool.c:
8072 address-pool: Add unicast addresses
8074 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8077 * gst/rtsp-server/rtsp-server.c:
8078 * tests/check/gst/rtspserver.c:
8079 rtsp-server: Limit the number of threads per server instance
8080 If we exceed the maximum, just round robin the clients over the existing
8083 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
8085 * gst/rtsp-server/rtsp-server.c:
8086 rtsp-server: No need to store the GMainContext in the client context
8088 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
8090 * tests/check/gst/rtspserver.c:
8091 tests: Add test for client disconnection
8093 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8095 * tests/check/gst/rtspserver.c:
8096 tests: Test client and session timeouts with multiple threads
8098 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
8100 * gst/rtsp-server/rtsp-address-pool.c:
8101 * gst/rtsp-server/rtsp-auth.c:
8102 * gst/rtsp-server/rtsp-client.c:
8103 * gst/rtsp-server/rtsp-media-factory-uri.c:
8104 * gst/rtsp-server/rtsp-media-factory.c:
8105 * gst/rtsp-server/rtsp-media.c:
8106 * gst/rtsp-server/rtsp-mount-points.c:
8107 * gst/rtsp-server/rtsp-server.c:
8108 * gst/rtsp-server/rtsp-session-media.c:
8109 * gst/rtsp-server/rtsp-session-pool.c:
8110 * gst/rtsp-server/rtsp-session.c:
8111 Document locking and its order
8113 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
8115 * tests/check/gst/rtspserver.c:
8116 tests: Test that slow DESCRIBE don't block other clients
8118 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
8120 * tests/check/gst/client.c:
8121 tests: Add tests for client-requested multicast address
8123 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
8125 * docs/libs/gst-rtsp-server-sections.txt:
8126 docs: Put the various functions in the right sections
8128 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
8130 * docs/libs/gst-rtsp-server-docs.sgml:
8131 * docs/libs/gst-rtsp-server-sections.txt:
8132 * gst/rtsp-server/rtsp-address-pool.c:
8133 * gst/rtsp-server/rtsp-address-pool.h:
8134 docs: Generate docs for GstRTSPAddressPool
8136 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8138 * gst/rtsp-server/rtsp-client.c:
8139 * gst/rtsp-server/rtsp-stream.c:
8140 * gst/rtsp-server/rtsp-stream.h:
8141 client: Check client provided addresses against the address pool
8143 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
8145 * gst/rtsp-server/rtsp-address-pool.c:
8146 * gst/rtsp-server/rtsp-address-pool.h:
8147 * tests/check/gst/addresspool.c:
8148 address-pool: Add API to request a specific address from the pool
8149 Also add relevant unit tests.
8151 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
8153 * tests/check/gst/mediafactory.c:
8154 tests: Check the passing around of a RTSPAddressPool
8155 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
8156 way down to the stream.
8158 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
8160 * tests/check/gst/addresspool.c:
8161 tests: Add more tests for the address pool
8163 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
8165 * gst/rtsp-server/rtsp-address-pool.c:
8166 address-pool: Fix off by one error
8167 When splitting a port range, the port after a skip is not part of range.
8169 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
8172 Automatic update of common submodule
8173 From 2de221c to 04c7a1e
8175 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
8178 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
8179 AM_CONFIG_HEADER was removed in automake 1.13
8180 https://bugzilla.gnome.org/show_bug.cgi?id=693368
8182 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
8185 Automatic update of common submodule
8186 From a942293 to 2de221c
8188 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8190 * gst/rtsp-server/rtsp-client.c:
8191 client: make sure the watch exists while sending data
8192 Protect the send_func with a lock. This allows us to wait for sending
8193 to complete before changing the send_func and user_data. We add an
8194 extra ref to the watch to make sure that it remains valid during
8196 When closing the connection, set the send_func to NULL
8197 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
8199 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8201 * tests/check/Makefile.am:
8202 tests: use GST_*_1_0 environment variables everywhere
8203 The _1_0 suffixed environment variables override the
8204 non-suffixed ones, so if we're in an environment that
8205 sets the _1_0 suffixed ones, such as jhbuild, we need
8206 to set those to make sure ours actually always get
8209 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8212 Automatic update of common submodule
8213 From acb04d9 to a942293
8215 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8217 * gst/rtsp-server/rtsp-client.c:
8218 rtsp-client: set the client backlog
8219 Set the client backlog to a reasonable default
8221 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
8223 * gst/rtsp-server/rtsp-media.c:
8224 rtsp-media: Make the element a constructor parameter
8225 https://bugzilla.gnome.org/show_bug.cgi?id=689594
8227 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8229 * docs/libs/Makefile.am:
8230 docs: Link with gcov library when gcov is enabled
8231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
8233 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8235 * gst/rtsp-server/rtsp-media.c:
8236 media: match prepare with unprepare
8237 Really unprepare when there were an equal amount of prepare calls.
8239 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8241 * gst/rtsp-server/rtsp-media.c:
8242 media: media has to be unprepared in finalize
8243 Because unprepare takes away the last ref on the media.
8245 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8247 * gst/rtsp-server/rtsp-client.c:
8248 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
8249 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
8250 We can't use the refcount to trigger unprepare because it is the unprepare call
8251 that removes the last refcount after all messages are consumed. What we should
8252 probably do is make a prepared refcount and only unprepare when the refcount
8255 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8257 * gst/rtsp-server/rtsp-media.c:
8258 media: let the source unref the last media ref
8259 the last ref to the media is held by the source so we don't need to add more ref
8260 and unrefs, we simply destroy the media when the source is gone.
8262 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8264 * gst/rtsp-server/rtsp-media.c:
8265 media: improve debug
8267 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * gst/rtsp-server/rtsp-media.c:
8271 Make sure we are in the right state when collecting the position and duration.
8272 Only make ourselves PREPARED when we were previously PREPARING.
8274 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8276 * gst/rtsp-server/rtsp-media.c:
8277 media: use g_object_ref/unref for GObjects
8279 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
8281 * gst/rtsp-server/rtsp-client.c:
8282 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
8283 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
8284 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
8285 isn't being used anymore.
8287 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
8289 * gst/rtsp-server/rtsp-media.c:
8290 Fix compiler warning
8292 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
8294 * gst/rtsp-server/rtsp-media-factory-uri.c:
8295 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
8297 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8299 * gst/rtsp-server/rtsp-session-media.h:
8302 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8304 * gst/rtsp-server/rtsp-media.c:
8305 * tests/check/gst/media.c:
8306 media: avoid element leak
8308 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8310 * gst/rtsp-server/rtsp-media.c:
8311 media: require an element in media constructor
8313 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8315 * gst/rtsp-server/rtsp-client.c:
8316 Revert "client: TEARDOWN brings that state to Init again"
8317 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
8318 The object is already disposed, there is no point in setting the state.
8320 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8322 * gst/rtsp-server/rtsp-client.c:
8323 client: TEARDOWN brings that state to Init again
8325 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8327 * docs/libs/gst-rtsp-server-sections.txt:
8328 * examples/test-auth.c:
8329 * gst/rtsp-server/rtsp-auth.c:
8330 * gst/rtsp-server/rtsp-auth.h:
8331 * gst/rtsp-server/rtsp-client.c:
8332 * gst/rtsp-server/rtsp-client.h:
8333 * gst/rtsp-server/rtsp-media-factory-uri.c:
8334 * gst/rtsp-server/rtsp-media-factory-uri.h:
8335 * gst/rtsp-server/rtsp-media-factory.c:
8336 * gst/rtsp-server/rtsp-media-factory.h:
8337 * gst/rtsp-server/rtsp-media.c:
8338 * gst/rtsp-server/rtsp-media.h:
8339 * gst/rtsp-server/rtsp-mount-points.c:
8340 * gst/rtsp-server/rtsp-mount-points.h:
8341 * gst/rtsp-server/rtsp-sdp.c:
8342 * gst/rtsp-server/rtsp-server.c:
8343 * gst/rtsp-server/rtsp-server.h:
8344 * gst/rtsp-server/rtsp-session-media.c:
8345 * gst/rtsp-server/rtsp-session-media.h:
8346 * gst/rtsp-server/rtsp-session-pool.c:
8347 * gst/rtsp-server/rtsp-session-pool.h:
8348 * gst/rtsp-server/rtsp-session.c:
8349 * gst/rtsp-server/rtsp-session.h:
8350 * gst/rtsp-server/rtsp-stream-transport.c:
8351 * gst/rtsp-server/rtsp-stream-transport.h:
8352 * gst/rtsp-server/rtsp-stream.c:
8353 * gst/rtsp-server/rtsp-stream.h:
8354 * tests/check/gst/media.c:
8355 rtsp: make object details private
8356 Make all object details private
8357 Add methods to access private bits
8359 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8361 * tests/check/Makefile.am:
8362 * tests/check/gst/media.c:
8363 tests: add media tests
8365 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8367 * gst/rtsp-server/rtsp-media.c:
8368 media: check if prepared for some methods
8369 Check that the media object is prepared before doing seek and getting the
8370 current position etc.
8371 Add some g_return checks.
8373 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8375 * tests/check/Makefile.am:
8376 * tests/check/gst/mediafactory.c:
8377 tests: add mediafactory test
8379 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8381 * gst/rtsp-server/rtsp-stream.c:
8382 stream: improve debug
8384 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8386 * gst/rtsp-server/rtsp-media.c:
8387 * gst/rtsp-server/rtsp-media.h:
8388 media: unref pipeline in finalize to avoid leaking it
8390 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8392 * gst/rtsp-server/rtsp-media-factory-uri.c:
8393 * gst/rtsp-server/rtsp-media.c:
8394 rtsp: use gst_object_unref on GstObjects
8396 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8398 * gst/rtsp-server/rtsp-media-factory.c:
8399 media-factory: require an url
8401 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8403 * examples/test-uri.c:
8404 examples: fix include
8406 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8408 * gst/rtsp-server/rtsp-server.h:
8409 server: remove unused include
8411 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * tests/check/Makefile.am:
8414 * tests/check/gst/mountpoints.c:
8415 tests: add test for mountpoints
8417 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8419 * gst/rtsp-server/rtsp-client.c:
8420 client: fix factory leak
8421 Keep the factory in the state object only for authorization checks and make
8422 sure we unref it on failure. Also don't keep invalid objects in the state
8425 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8427 * gst/rtsp-server/rtsp-mount-points.c:
8428 mounts: add g_return_if guards
8430 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8432 * tests/check/gst/client.c:
8433 tests: add more tests
8435 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8437 * gst/rtsp-server/rtsp-client.c:
8438 client: improve debug
8440 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-client.c:
8443 client: improve debug and fix leaks
8444 Cleanup the uri and session when there is a bad request.
8446 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8453 * tests/check/gst/client.c:
8454 test: add test for session in options request
8456 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8458 * gst/rtsp-server/rtsp-client.c:
8459 client: use 454 when session can't be found
8460 We should use 454 when a session can't be found because there was no session
8461 pool configured in the server. This is not a server configuration problem
8462 because the server on which the request is done might not be the same one that
8463 will keep the sessions for us and so it does not need to support sessions.
8465 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-client.c:
8468 client: only free connection when there is one
8469 It's possible that the client doesn't have a connection when we try to free it.
8471 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8473 * tests/check/Makefile.am:
8474 * tests/check/gst/client.c:
8475 tests: add unit test for the client object
8477 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8479 * gst/rtsp-server/rtsp-client.c:
8480 client: small cleanup
8482 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/rtsp-client.h:
8485 client: remove unused include
8487 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8489 * gst/rtsp-server/rtsp-client.c:
8490 client: fix compilation
8492 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8494 * gst/rtsp-server/rtsp-client.c:
8495 client: call destroy without the lock
8497 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8499 * gst/rtsp-server/rtsp-client.c:
8500 * gst/rtsp-server/rtsp-client.h:
8501 client: make the client usable without a socket
8502 Make a method to let the client handle a message and a callback when the client
8503 wants us to send a response message back. This makes it possible to also use the
8504 client object without the sockets, which should make it easier to test.
8506 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8508 * gst/rtsp-server/rtsp-client.c:
8509 * gst/rtsp-server/rtsp-client.h:
8510 client: small cleanup
8512 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8514 * docs/libs/gst-rtsp-server-sections.txt:
8515 * gst/rtsp-server/rtsp-client.c:
8516 * gst/rtsp-server/rtsp-client.h:
8517 * gst/rtsp-server/rtsp-server.c:
8518 client: remove reference to server
8519 We don't need to keep a ref to the server
8521 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8523 * gst/rtsp-server/rtsp-client.c:
8524 * gst/rtsp-server/rtsp-client.h:
8526 Also add some g_return_if()
8528 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8530 * gst/rtsp-server/rtsp-client.c:
8531 client: log more errors
8533 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8535 * gst/rtsp-server/rtsp-client.c:
8536 client: fix compilation
8538 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-client.c:
8541 * gst/rtsp-server/rtsp-client.h:
8542 client: add generic close-after-send support
8543 Add a property to send_response() to close the connection after the response has
8544 been sent to the client.
8546 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8549 * docs/libs/gst-rtsp-server-docs.sgml:
8550 * docs/libs/gst-rtsp-server-sections.txt:
8551 * docs/libs/gst-rtsp-server.types:
8552 * examples/test-auth.c:
8553 * examples/test-launch.c:
8554 * examples/test-mp4.c:
8555 * examples/test-multicast.c:
8556 * examples/test-multicast2.c:
8557 * examples/test-ogg.c:
8558 * examples/test-readme.c:
8559 * examples/test-sdp.c:
8560 * examples/test-uri.c:
8561 * examples/test-video.c:
8562 * gst/rtsp-server/Makefile.am:
8563 * gst/rtsp-server/rtsp-auth.h:
8564 * gst/rtsp-server/rtsp-client.c:
8565 * gst/rtsp-server/rtsp-client.h:
8566 * gst/rtsp-server/rtsp-media-mapping.c:
8567 * gst/rtsp-server/rtsp-media-mapping.h:
8568 * gst/rtsp-server/rtsp-mount-points.c:
8569 * gst/rtsp-server/rtsp-mount-points.h:
8570 * gst/rtsp-server/rtsp-server.c:
8571 * gst/rtsp-server/rtsp-server.h:
8572 * gst/rtsp-server/rtsp-session-media.c:
8573 * gst/rtsp-server/rtsp-session-pool.c:
8574 * gst/rtsp-server/rtsp-session-pool.h:
8575 * tests/check/gst/rtspserver.c:
8576 MediaMapping -> MountPoints
8577 Describes better what the object manages.
8579 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8582 configure: bump required version of -base
8584 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/rtsp-media.c:
8589 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * gst/rtsp-server/rtsp-media.c:
8592 * gst/rtsp-server/rtsp-media.h:
8593 media: support more Range formats
8594 Use the new -base methods to convert the Range string into a seek start and stop
8597 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8599 * examples/test-launch.c:
8600 examples: fix whitespace
8602 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8604 * examples/test-auth.c:
8605 test-auth: add example of how to remove sessions
8606 Add an example of the session filter api.
8608 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8610 * examples/test-uri.c:
8611 test-uri: remove mapping example
8613 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8615 * examples/test-uri.c:
8616 test-uri: fix callback signature
8618 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8620 * gst/rtsp-server/rtsp-media-factory.c:
8621 factory: keep ref to factory while media active
8622 While the media from a factory is alive, keep a ref to the factory.
8623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
8625 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8627 * gst/rtsp-server/rtsp-media-factory-uri.c:
8628 factory-uri: add some debug
8630 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8632 * gst/rtsp-server/rtsp-stream.c:
8633 stream: set udp sources to PLAYING
8634 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
8635 so that it doesn't cause our pipeline to produce ASYNC-DONE.
8637 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8639 * gst/rtsp-server/rtsp-media-factory-uri.c:
8640 factory-uri: take ref to factory
8641 Take a ref to the factory that we place in our list.
8643 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8645 * tests/Makefile.am:
8646 * tests/test-reuse.c:
8647 test: add test for server reuse
8648 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
8650 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
8652 * gst/rtsp-server/rtsp-server.c:
8653 server: start and stop multiple times
8654 Stop listening on the RTSP port when the GSource is removed, so clients
8655 can't connect and the server can be started again.
8656 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
8658 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8660 * gst/rtsp-server/rtsp-server.c:
8661 server: fix small leak
8663 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8665 * gst/rtsp-server/rtsp-media.c:
8666 media: unref source in finish_unprepare
8667 The source is created in prepare, unref it in finish_unprepare.
8668 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
8670 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
8672 * gst/rtsp-server/rtsp-client.c:
8673 * gst/rtsp-server/rtsp-media.c:
8674 rtsp-media: remove bus watch before finalizing
8675 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
8676 * An extra media ref is added for the bus watch. This extra ref is unreffed by
8677 the GDestroyNotify function.
8678 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
8679 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
8680 gst_rtsp_media_unprepare before unreffing the media.
8681 This way, the bus watch will be removed before the media is finalized.
8682 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
8684 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
8686 * gst/rtsp-server/rtsp-client.c:
8687 * gst/rtsp-server/rtsp-client.h:
8688 client: wait until the TEARDOWN response is sent to close the connection
8689 Responses can be sent async so we need to wait until the TEARDOWN response has
8690 been written before we close the connection to the client. This avoids the risk
8691 of writing/polling closed sockets.
8692 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
8694 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
8696 * gst/rtsp-server/rtsp-stream.c:
8697 rtsp-stream: plug socket leak
8698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
8700 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
8703 Automatic update of common submodule
8704 From 6bb6951 to a72faea
8706 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
8708 * gst/rtsp-server/rtsp-media-factory-uri.c:
8709 rtsp-server: don't use deprecated API
8711 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
8713 * gst/rtsp-server/rtsp-client.c:
8714 rtsp-client: fix unused-but-set-variable compiler warning
8715 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
8717 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8720 * docs/libs/gst-rtsp-server-sections.txt:
8721 * gst/rtsp-server/rtsp-client.c:
8724 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8726 * examples/Makefile.am:
8727 * examples/test-multicast2.c:
8728 examples: add another multicast example
8729 Add an example for how to configure separate multicast ranges for each media
8732 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8734 * examples/test-multicast.c:
8737 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8739 * gst/rtsp-server/rtsp-client.c:
8740 * gst/rtsp-server/rtsp-media.c:
8741 * gst/rtsp-server/rtsp-session-media.c:
8742 * gst/rtsp-server/rtsp-session-media.h:
8743 * gst/rtsp-server/rtsp-stream-transport.c:
8744 * gst/rtsp-server/rtsp-stream-transport.h:
8745 stream: use the address managed by the stream
8746 Use the address managed by the stream for multicast. This allows us to have 1
8747 multicast address for each stream.
8748 Because the address is now managed by the stream we don't have to pass it around
8750 Set the address pool on the streams.
8752 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8754 * gst/rtsp-server/rtsp-client.c:
8755 * gst/rtsp-server/rtsp-media.c:
8756 * gst/rtsp-server/rtsp-stream.c:
8759 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8761 * gst/rtsp-server/rtsp-media.c:
8762 * gst/rtsp-server/rtsp-media.h:
8763 media: add signal for new streams
8764 This allows applications to listen for new streams and configure properties on
8765 them, like the address pool.
8767 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8769 * gst/rtsp-server/rtsp-media.c:
8770 media: configure address pool in new streams
8772 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8774 * gst/rtsp-server/rtsp-stream.c:
8775 * gst/rtsp-server/rtsp-stream.h:
8776 stream: add methods to deal with address pool
8777 Add methods to get and set the address pool for the stream
8778 Add method to allocate and get the multicast addresses for this stream.
8780 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8782 * docs/libs/gst-rtsp-server-sections.txt:
8783 * gst/rtsp-server/rtsp-media.c:
8784 * gst/rtsp-server/rtsp-media.h:
8785 media: remove MTU property
8786 It is a stream property
8788 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8790 * gst/rtsp-server/rtsp-client.c:
8791 client: set blocksize only on stream
8792 Set the blocksize only on the current stream.
8794 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8796 * gst/rtsp-server/rtsp-stream.c:
8797 stream: share src and sink sockets
8798 the allocated socket is in the used-socket property, not socket.
8800 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8802 * gst/rtsp-server/rtsp-address-pool.c:
8803 * gst/rtsp-server/rtsp-address-pool.h:
8804 * gst/rtsp-server/rtsp-client.c:
8805 * gst/rtsp-server/rtsp-session-media.c:
8806 * gst/rtsp-server/rtsp-session-media.h:
8807 * gst/rtsp-server/rtsp-stream-transport.c:
8808 * gst/rtsp-server/rtsp-stream-transport.h:
8809 * tests/check/gst/addresspool.c:
8810 rtsp: make address-pool return an address object
8811 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
8812 store more info in the structure and allows us to more easily return the address
8813 to the right pool when no longer needed.
8814 Pass the address to the StreamTransport so that we can return it to the pool
8815 when the stream transport is freed or changed.
8817 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8819 * examples/Makefile.am:
8820 * examples/test-multicast.c:
8821 examples: add multicast example
8822 Show how to set up the multicast address pool so that media can be
8823 server with multicast.
8825 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8827 * gst/rtsp-server/rtsp-client.c:
8828 * gst/rtsp-server/rtsp-media-factory.c:
8829 * gst/rtsp-server/rtsp-media-factory.h:
8830 * gst/rtsp-server/rtsp-media.c:
8831 * gst/rtsp-server/rtsp-media.h:
8832 rtsp: use AddressPool
8833 Remove the multicast_group property.
8834 Use the configured addresspool to allocate multicast addresses.
8836 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8838 * gst/rtsp-server/rtsp-address-pool.c:
8839 * gst/rtsp-server/rtsp-address-pool.h:
8840 address-pool: add clear method
8842 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-address-pool.c:
8845 address-pool: small cleanups
8847 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8849 * tests/check/Makefile.am:
8850 * tests/check/gst/addresspool.c:
8851 tests: add addresspool unit test
8853 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8855 * gst/rtsp-server/Makefile.am:
8856 * gst/rtsp-server/rtsp-address-pool.c:
8857 * gst/rtsp-server/rtsp-address-pool.h:
8858 address-pool: add object to manage multicast addresses
8859 Make an object that can manage a rage of multicast addresses and ports.
8861 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8863 * gst/rtsp-server/rtsp-server.c:
8864 server: set default max-threads property
8866 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8868 * gst/rtsp-server/rtsp-media.c:
8869 media: wait for concurrent _prepare
8870 If a prepare is busy, wait for the result.
8872 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8874 * gst/rtsp-server/rtsp-media.c:
8875 media: add lock around message handler
8876 We don't want to dispatch messages while we are still processing the result of
8879 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8881 * gst/rtsp-server/rtsp-media.c:
8882 * gst/rtsp-server/rtsp-media.h:
8883 media: add lock to protect state changes
8885 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8887 * gst/rtsp-server/rtsp-stream.c:
8888 * gst/rtsp-server/rtsp-stream.h:
8891 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8893 * gst/rtsp-server/rtsp-stream-transport.c:
8894 * gst/rtsp-server/rtsp-stream-transport.h:
8895 * gst/rtsp-server/rtsp-stream.c:
8896 stream-transport: add keep-alive method
8898 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8900 * gst/rtsp-server/rtsp-stream-transport.c:
8901 * gst/rtsp-server/rtsp-stream-transport.h:
8902 * gst/rtsp-server/rtsp-stream.c:
8903 stream-transport: add method to handle RTP/RTCP
8904 Call new methods instead of poking into the structures directly.
8906 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8908 * gst/rtsp-server/rtsp-session-media.c:
8909 * gst/rtsp-server/rtsp-session-media.h:
8910 session-media: add locking
8912 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8914 * gst/rtsp-server/rtsp-session.c:
8915 * gst/rtsp-server/rtsp-session.h:
8916 session: add locking
8918 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8920 * gst/rtsp-server/rtsp-server.c:
8921 server: free old socket
8923 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8925 * gst/rtsp-server/rtsp-media-mapping.c:
8926 * gst/rtsp-server/rtsp-media-mapping.h:
8927 mapping: add locking
8929 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8931 * gst/rtsp-server/rtsp-media-factory.c:
8932 media-factory: add locking
8934 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8936 * gst/rtsp-server/rtsp-auth.c:
8937 * gst/rtsp-server/rtsp-auth.h:
8940 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8942 * gst/rtsp-server/rtsp-server.c:
8943 * gst/rtsp-server/rtsp-server.h:
8944 server: add max-thread property
8946 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8948 * gst/rtsp-server/rtsp-server.c:
8949 * gst/rtsp-server/rtsp-server.h:
8950 server: use a threadpool for the mainloops
8952 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8954 * gst/rtsp-server/rtsp-client.c:
8955 * gst/rtsp-server/rtsp-client.h:
8956 client: rename method
8957 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
8958 don't really create the client from the socket, we use the socket for the
8961 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8963 * gst/rtsp-server/rtsp-client.c:
8964 * gst/rtsp-server/rtsp-client.h:
8965 * gst/rtsp-server/rtsp-server.c:
8966 server: rework maincontext handling in clients
8967 Make a separate method to attach a client to a MainContext.
8968 Let the server decide in what GMainContext the client will operate and give this
8969 context to the client in attach. Then the server can later decide to use a
8970 separate thread for each client or just use the mainthread.
8972 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8974 * gst/rtsp-server/rtsp-client.c:
8975 * gst/rtsp-server/rtsp-session.c:
8976 * gst/rtsp-server/rtsp-session.h:
8977 session: move session header code in session object
8979 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
8983 * examples/test-auth.c:
8984 * examples/test-launch.c:
8985 * examples/test-mp4.c:
8986 * examples/test-ogg.c:
8987 * examples/test-readme.c:
8988 * examples/test-sdp.c:
8989 * examples/test-uri.c:
8990 * examples/test-video.c:
8991 * gst/rtsp-server/rtsp-auth.c:
8992 * gst/rtsp-server/rtsp-auth.h:
8993 * gst/rtsp-server/rtsp-client.c:
8994 * gst/rtsp-server/rtsp-client.h:
8995 * gst/rtsp-server/rtsp-media-factory-uri.c:
8996 * gst/rtsp-server/rtsp-media-factory-uri.h:
8997 * gst/rtsp-server/rtsp-media-factory.c:
8998 * gst/rtsp-server/rtsp-media-factory.h:
8999 * gst/rtsp-server/rtsp-media-mapping.c:
9000 * gst/rtsp-server/rtsp-media-mapping.h:
9001 * gst/rtsp-server/rtsp-media.c:
9002 * gst/rtsp-server/rtsp-media.h:
9003 * gst/rtsp-server/rtsp-params.c:
9004 * gst/rtsp-server/rtsp-params.h:
9005 * gst/rtsp-server/rtsp-sdp.c:
9006 * gst/rtsp-server/rtsp-sdp.h:
9007 * gst/rtsp-server/rtsp-server.c:
9008 * gst/rtsp-server/rtsp-server.h:
9009 * gst/rtsp-server/rtsp-session-media.c:
9010 * gst/rtsp-server/rtsp-session-media.h:
9011 * gst/rtsp-server/rtsp-session-pool.c:
9012 * gst/rtsp-server/rtsp-session-pool.h:
9013 * gst/rtsp-server/rtsp-session.c:
9014 * gst/rtsp-server/rtsp-session.h:
9015 * gst/rtsp-server/rtsp-stream-transport.c:
9016 * gst/rtsp-server/rtsp-stream-transport.h:
9017 * gst/rtsp-server/rtsp-stream.c:
9018 * gst/rtsp-server/rtsp-stream.h:
9019 * tests/check/gst/rtspserver.c:
9020 * tests/test-cleanup.c:
9023 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9025 * gst/rtsp-server/rtsp-media.c:
9026 * gst/rtsp-server/rtsp-session-media.c:
9027 * gst/rtsp-server/rtsp-session.c:
9028 rtsp-server: added annotations to indicate type of ownership transfer of return values
9029 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9031 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
9034 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
9036 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
9039 * bindings/Makefile.am:
9040 * bindings/vala/Makefile.am:
9041 * bindings/vala/gst-rtsp-server-0.10.deps:
9042 * bindings/vala/gst-rtsp-server-0.10.vapi:
9043 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9044 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9045 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9046 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9047 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9049 bindings: remove vala bindings
9050 They'll be reunited with the other GStreamer bindings
9051 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9053 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9055 * gst/rtsp-server/rtsp-client.c:
9056 * gst/rtsp-server/rtsp-session-media.c:
9057 * gst/rtsp-server/rtsp-session-media.h:
9058 * gst/rtsp-server/rtsp-stream-transport.c:
9059 * gst/rtsp-server/rtsp-stream-transport.h:
9060 rtsp: only create transport when needed
9061 Only create the StreamTransport when configured.
9063 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9065 * gst/rtsp-server/rtsp-client.c:
9066 client: small cleanup
9068 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9070 * gst/rtsp-server/rtsp-client.c:
9071 * gst/rtsp-server/rtsp-client.h:
9072 * gst/rtsp-server/rtsp-stream-transport.c:
9073 * gst/rtsp-server/rtsp-stream-transport.h:
9074 rtsp: refactor configuration of transport
9075 Move the configuration of the transport to a place where it makes
9078 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9080 * gst/rtsp-server/rtsp-client.c:
9081 client: refactor transport parsing
9083 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9085 * gst/rtsp-server/rtsp-client.c:
9086 client: refuse to change the MTU on shared media
9087 If we change the MTU of chared media, it changes for all clients.
9088 We don't want to set the MTU to something large for clients that
9091 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9093 * examples/test-mp4.c:
9094 * gst/rtsp-server/rtsp-media.c:
9095 small fixes to docs and debug
9097 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9099 * gst/rtsp-server/rtsp-stream.c:
9100 stream: transports must already have been removed
9102 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9104 * gst/rtsp-server/rtsp-media.c:
9105 * gst/rtsp-server/rtsp-stream.c:
9106 * gst/rtsp-server/rtsp-stream.h:
9107 stream: improve join and leave of the pipeline
9109 Do the cleanup properly
9112 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9114 * gst/rtsp-server/rtsp-media.c:
9115 media: move unprepare below default implementation
9116 Makes it easier to find the default implementation
9118 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9120 * gst/rtsp-server/rtsp-media.c:
9121 media: signal unprepared when we actually finish
9123 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * gst/rtsp-server/rtsp-media.c:
9126 media: no need to unlock, unprepare does that when needed
9128 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9130 * docs/libs/gst-rtsp-server-sections.txt:
9131 * gst/rtsp-server/rtsp-media-factory.h:
9132 * gst/rtsp-server/rtsp-media-mapping.c:
9133 * gst/rtsp-server/rtsp-media.h:
9134 * gst/rtsp-server/rtsp-params.c:
9135 * gst/rtsp-server/rtsp-server.c:
9136 * gst/rtsp-server/rtsp-session-pool.h:
9137 * gst/rtsp-server/rtsp-session.c:
9138 * gst/rtsp-server/rtsp-session.h:
9139 * gst/rtsp-server/rtsp-stream-transport.h:
9140 * gst/rtsp-server/rtsp-stream.h:
9143 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9145 * gst/rtsp-server/rtsp-client.c:
9146 * gst/rtsp-server/rtsp-media-mapping.h:
9147 * gst/rtsp-server/rtsp-media.c:
9148 * gst/rtsp-server/rtsp-media.h:
9149 * gst/rtsp-server/rtsp-server.h:
9150 * gst/rtsp-server/rtsp-stream.c:
9151 * gst/rtsp-server/rtsp-stream.h:
9152 rtsp: fix MTU setting
9153 Fix setting of the MTU. There is no need for a vmethod.
9155 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9160 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9163 configure: bump version number after refactoring
9165 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9167 * gst/rtsp-server/Makefile.am:
9168 * gst/rtsp-server/rtsp-client.c:
9169 * gst/rtsp-server/rtsp-client.h:
9170 * gst/rtsp-server/rtsp-media-factory-uri.c:
9171 * gst/rtsp-server/rtsp-media-factory.c:
9172 * gst/rtsp-server/rtsp-media-factory.h:
9173 * gst/rtsp-server/rtsp-media.c:
9174 * gst/rtsp-server/rtsp-media.h:
9175 * gst/rtsp-server/rtsp-sdp.c:
9176 * gst/rtsp-server/rtsp-session-media.c:
9177 * gst/rtsp-server/rtsp-session-media.h:
9178 * gst/rtsp-server/rtsp-session.c:
9179 * gst/rtsp-server/rtsp-session.h:
9180 * gst/rtsp-server/rtsp-stream-transport.c:
9181 * gst/rtsp-server/rtsp-stream-transport.h:
9182 * gst/rtsp-server/rtsp-stream.c:
9183 * gst/rtsp-server/rtsp-stream.h:
9184 rtsp: massive refactoring
9185 Make GObjects from the remaining simple structures.
9186 Remove GstRTSPSessionStream, it's not needed.
9187 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
9188 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
9189 a GstRTSPStream should be transported to a client.
9190 Rename GstRTSPMediaFactory::get_element -> create_element because that
9191 more accurately describes what it does.
9192 Make nice methods instead of poking in the structures.
9193 Move some methods inside the relevant object source code.
9194 Use GPtrArray to store objects instead of plain arrays, it is more
9195 natural and allows us to more easily clean up.
9196 Move the allocation of udp ports to the Stream object. The Stream object
9197 contains the elements needed to stream the media to a client.
9198 Improve the prepare and unprepare methods. Unprepare should now undo
9199 everything prepare did. Improve also async unprepare when doing EOS on
9200 shutdown. Make sure we always unprepare correctly.
9202 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
9204 * gst/rtsp-server/rtsp-client.c:
9205 rtsp-client: Unref server address clients connected to
9206 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
9208 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
9210 * gst/rtsp-server/rtsp-server.c:
9211 rtsp-server: don't ref server socket if it is NULL
9212 Fixes test_bind_already_in_use unit test again after commit 6a497440.
9213 https://bugzilla.gnome.org/show_bug.cgi?id=686644
9215 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
9217 * tests/check/Makefile.am:
9218 tests: Add libgio link dependency
9219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
9221 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9223 * gst/rtsp-server/rtsp-media-mapping.c:
9224 * gst/rtsp-server/rtsp-media-mapping.h:
9225 rtsp-media-mapping: rename find_media vfunc to find_factory
9226 The virtual method and class method should have the same name
9227 so it is correctly represented in GIR file
9228 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9230 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9232 * gst/rtsp-server/rtsp-auth.c:
9233 * gst/rtsp-server/rtsp-client.c:
9234 * gst/rtsp-server/rtsp-media-factory-uri.c:
9235 * gst/rtsp-server/rtsp-media-factory.c:
9236 * gst/rtsp-server/rtsp-media-mapping.c:
9237 * gst/rtsp-server/rtsp-media.c:
9238 * gst/rtsp-server/rtsp-server.c:
9239 * gst/rtsp-server/rtsp-session-pool.c:
9240 * gst/rtsp-server/rtsp-session.c:
9241 rtsp-server: fixed comments and GIR annotations
9242 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9244 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
9246 * gst/rtsp-server/rtsp-media-mapping.c:
9247 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
9249 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
9251 * gst/rtsp-server/rtsp-server.c:
9252 rtsp-server: allow binding on port 0 (binds on a random port)
9254 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
9256 * gst/rtsp-server/rtsp-server.c:
9257 * gst/rtsp-server/rtsp-server.h:
9258 rtsp-server: add bound-port property
9259 bound-port can be used to retrieve the port number when the server is bound on
9260 port 0, which binds on a random port.
9262 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
9264 * gst/rtsp-server/rtsp-media-factory.c:
9265 * gst/rtsp-server/rtsp-media-factory.h:
9266 rtsp-media-factory: make ::get_element overridable by GI bindings
9267 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
9268 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
9269 as the invoker for ::get_element(), making it overridable by GI generated
9272 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9274 * gst/rtsp-server/rtsp-media-factory-uri.c:
9275 rtsp-media-factory-uri: don't autoplug parsers in a loop
9276 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
9279 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9281 * gst/rtsp-server/Makefile.am:
9282 Explicitly link against gio. Fix link error on mac.
9284 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9286 * gst/rtsp-server/rtsp-session.c:
9287 session: add ttl to the transport header in SETUP
9288 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
9290 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9292 * gst/rtsp-server/rtsp-client.c:
9293 * gst/rtsp-server/rtsp-client.h:
9294 * gst/rtsp-server/rtsp-media.c:
9295 client: Use client transport settings for multicast if allowed.
9296 This patch makes it possible for the client to send transport settings for
9297 multicast (destination && ttl). Client settings must be explicitly allowed or
9298 the server will use its own settings.
9299 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
9301 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
9304 Automatic update of common submodule
9305 From 6c0b52c to 6bb6951
9307 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
9309 * gst/rtsp-server/rtsp-client.c:
9310 rtsp-client: do not destroy the rtsp watch
9311 Don't destroy the client watch while dispatching. The rtsp watch is
9312 automatically destroyed after the rtsp watch function closed() has
9314 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
9316 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9319 Automatic update of common submodule
9320 From 4f962f7 to 6c0b52c
9322 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
9324 * gst/rtsp-server/rtsp-media.c:
9325 media: fix check for seekability
9327 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9329 * gst/rtsp-server/rtsp-client.c:
9330 client: use more GIO
9331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
9333 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * gst/rtsp-server/rtsp-server.c:
9336 server: remove obsolete includes
9338 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9340 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
9341 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
9342 be available in "on_new_ssrc". The transports are added in
9343 gst_rtsp_media_set_state when going to PLAYING state. However,
9344 "on_new_ssrc" might be called before this happens.
9345 https://bugzilla.gnome.org/show_bug.cgi?id=683304
9347 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9349 * gst/rtsp-server/rtsp-client.c:
9350 * gst/rtsp-server/rtsp-client.h:
9351 rtsp-client: add signals for rtsp requests (fixes #683287)
9353 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9355 * gst/rtsp-server/rtsp-client.c:
9356 * gst/rtsp-server/rtsp-client.h:
9357 add new-session signal to rtsp-client (fixes #683058)
9359 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
9362 Automatic update of common submodule
9363 From 668acee to 4f962f7
9365 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
9367 * gst/rtsp-server/rtsp-server.c:
9368 * tests/check/gst/rtspserver.c:
9369 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
9370 Do not assume that *error is set in g_socket_address_enumerator_next.
9371 Added test_bind_already_in_use unit-test.
9372 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
9374 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
9377 Automatic update of common submodule
9378 From 94ccf4c to 668acee
9380 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
9382 * gst/rtsp-server/rtsp-client.c:
9383 * gst/rtsp-server/rtsp-client.h:
9384 rtsp-client: make create_sdp virtual method
9385 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
9387 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9390 Automatic update of common submodule
9391 From 98e386f to 94ccf4c
9393 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9395 * gst/rtsp-server/rtsp-client.c:
9398 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
9400 * gst/rtsp-server/rtsp-client.c:
9401 * gst/rtsp-server/rtsp-client.h:
9402 * gst/rtsp-server/rtsp-server.c:
9403 * gst/rtsp-server/rtsp-server.h:
9404 rtsp-server: use an existing socket to establish HTTP tunnel
9405 Make it possible to transfer a socket from an HTTP server to be used as
9406 an RTSP over HTTP tunnel.
9408 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
9410 * gst/rtsp-server/rtsp-client.c:
9411 * gst/rtsp-server/rtsp-media.c:
9412 * gst/rtsp-server/rtsp-media.h:
9413 rtsp: Handle the blocksize parameter
9414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
9416 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
9418 * tests/check/Makefile.am:
9419 * tests/check/gst/rtspserver.c:
9420 Have unit test get header from source dir, not installed dir
9421 This makes compilation of unit tests work in a build directory other
9422 than the source directory.
9423 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
9425 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
9427 * gst/rtsp-server/rtsp-media.c:
9428 rtsp-media: update for gst_element_make_from_uri() changes
9430 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
9433 * tests/Makefile.am:
9434 * tests/check/Makefile.am:
9435 * tests/check/gst/rtspserver.c:
9437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
9439 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
9441 * gst/rtsp-server/rtsp-media.c:
9442 rtsp-media: don't collect media stats when going to NULL
9443 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
9445 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9447 * gst/rtsp-server/rtsp-client.c:
9448 client: don't leak transports
9450 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
9452 * gst/rtsp-server/rtsp-client.c:
9453 rtsp-client: free transport on no_stream in SETUP handler
9455 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
9457 * gst/rtsp-server/rtsp-client.c:
9458 rtsp-client: changed session media iteration
9459 In client_unlink_session: now don't iterate in session->medias
9460 list where items are removed by gst_rtsp_session_release_media.
9461 Instead, repeatedly remove the first item.
9463 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
9465 * gst/rtsp-server/rtsp-client.c:
9466 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
9467 GstRTSPSessionMedia is not a GObject type. When the
9468 GstRTSPSession is freed, it will free the media.
9470 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
9472 * gst/rtsp-server/rtsp-media-factory.c:
9473 factory: plug pad leak in collect_streams
9474 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
9475 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
9476 will take one reference, and the other reference will otherwise
9479 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9482 configure: suppress some warnings when debug is disabled
9483 Warnings about unused variables should be suppressed if core has the
9484 debug system disabled.
9485 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9487 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9489 * docs/libs/Makefile.am:
9490 docs: fix build in uninstalled setup
9491 Include gst-plugins-base libs properly.
9493 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
9495 * docs/libs/gst-rtsp-server.types:
9496 docs: include headers defining rtsp-server object types
9497 Fixes compiler warnings during docs build.
9498 https://bugzilla.gnome.org/show_bug.cgi?id=676824
9500 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
9503 configure: Add warning flags for compiler when configuring
9504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9506 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9509 Automatic update of common submodule
9510 From 03a0e57 to 98e386f
9512 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9515 Automatic update of common submodule
9516 From 1fab359 to 03a0e57
9518 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
9520 * gst/rtsp-server/rtsp-client.c:
9521 client: fix GSocketAddress leak in gst_rtsp_client_accept
9522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
9524 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9527 Automatic update of common submodule
9528 From f1b5a96 to 1fab359
9530 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9533 Automatic update of common submodule
9534 From 92b7266 to f1b5a96
9536 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9539 Automatic update of common submodule
9540 From ec1c4a8 to 92b7266
9542 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9545 Automatic update of common submodule
9546 From 3429ba6 to ec1c4a8
9548 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
9550 * gst/rtsp-server/rtsp-auth.c:
9551 * gst/rtsp-server/rtsp-client.c:
9552 * gst/rtsp-server/rtsp-media-factory-uri.c:
9553 * gst/rtsp-server/rtsp-server.c:
9554 rtsp: fix compiler warnings
9555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
9557 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9560 Automatic update of common submodule
9561 From dc70203 to 3429ba6
9563 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9565 * gst/rtsp-server/rtsp-client.c:
9566 * gst/rtsp-server/rtsp-media-factory.c:
9567 * gst/rtsp-server/rtsp-media-factory.h:
9568 * gst/rtsp-server/rtsp-media.c:
9569 * gst/rtsp-server/rtsp-media.h:
9570 * gst/rtsp-server/rtsp-server.c:
9571 * gst/rtsp-server/rtsp-server.h:
9572 * gst/rtsp-server/rtsp-session-pool.c:
9573 * gst/rtsp-server/rtsp-session-pool.h:
9574 rtsp-server: port to new thread API
9576 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9579 Automatic update of common submodule
9580 From 6db25be to dc70203
9582 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9584 * gst/rtsp-server/rtsp-auth.c:
9585 * gst/rtsp-server/rtsp-auth.h:
9586 * gst/rtsp-server/rtsp-client.c:
9587 rtsp-server: Fix compilation and compiler warnings
9589 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9593 * gst/rtsp-server/Makefile.am:
9594 configure: Modernize autotools setup a bit
9595 Also we now only create tar.bz2 and tar.xz tarballs.
9597 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9600 Automatic update of common submodule
9601 From 464fe15 to 6db25be
9603 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9606 Automatic update of common submodule
9607 From 7fda524 to 464fe15
9609 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9612 * docs/libs/Makefile.am:
9613 * docs/version.entities.in:
9615 * gst/rtsp-server/Makefile.am:
9616 * pkgconfig/Makefile.am:
9617 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9618 * pkgconfig/gstreamer-rtsp-server.pc.in:
9619 * tests/Makefile.am:
9620 rtsp-server: Update versioning
9622 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9624 Merge remote-tracking branch 'origin/0.10'
9626 gst/rtsp-server/rtsp-session-pool.c
9628 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9630 * gst/rtsp-server/rtsp-session-pool.c:
9631 rtsp-server: Don't use deprecated GLib API
9633 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9635 Replace master with 0.11
9637 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9639 Merge branch 'master' into 0.11
9641 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 Merge branch 'master' into 0.11
9645 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
9648 A couple minor typo fixes
9650 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9652 * gst/rtsp-server/rtsp-media.c:
9653 media: fix state of the appqueue
9655 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9657 * gst/rtsp-server/rtsp-media-factory-uri.c:
9658 factory: use videoconvert
9660 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9662 * gst/rtsp-server/rtsp-media-factory-uri.c:
9663 factory: change to new style caps
9665 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9667 * gst/rtsp-server/rtsp-client.c:
9668 * gst/rtsp-server/rtsp-client.h:
9669 * gst/rtsp-server/rtsp-media-factory-uri.c:
9670 * gst/rtsp-server/rtsp-media.c:
9671 * gst/rtsp-server/rtsp-server.c:
9672 * gst/rtsp-server/rtsp-server.h:
9673 * gst/rtsp-server/rtsp-session-pool.c:
9674 rtsp-server: port to GIO
9677 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9680 configure: fix build
9682 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9685 docs: fix for gst_rtsp_server_set_port() -> _set_service()
9686 https://bugzilla.gnome.org/show_bug.cgi?id=666548
9688 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9691 * examples/Makefile.am:
9692 First rule of gst-rtsp-server club: don't talk about gst-phonon
9694 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9697 * pkgconfig/Makefile.am:
9698 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9699 * pkgconfig/gstreamer-rtsp-server.pc.in:
9700 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
9701 For consistency with all other modules.
9703 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9705 * gst/rtsp-server/rtsp-client.c:
9706 rtsp-client: update for new map API
9708 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9711 * bindings/Makefile.am:
9712 * bindings/python/Makefile.am:
9713 * bindings/python/arg-types.py:
9714 * bindings/python/codegen/Makefile.am:
9715 * bindings/python/codegen/__init__.py:
9716 * bindings/python/codegen/argtypes.py:
9717 * bindings/python/codegen/code-coverage.py:
9718 * bindings/python/codegen/codegen.py:
9719 * bindings/python/codegen/definitions.py:
9720 * bindings/python/codegen/defsparser.py:
9721 * bindings/python/codegen/docextract.py:
9722 * bindings/python/codegen/docgen.py:
9723 * bindings/python/codegen/fileprefix.override:
9724 * bindings/python/codegen/fileprefixmodule.c:
9725 * bindings/python/codegen/h2def.py:
9726 * bindings/python/codegen/mergedefs.py:
9727 * bindings/python/codegen/mkskel.py:
9728 * bindings/python/codegen/override.py:
9729 * bindings/python/codegen/reversewrapper.py:
9730 * bindings/python/codegen/scmexpr.py:
9731 * bindings/python/rtspserver-types.defs:
9732 * bindings/python/rtspserver.defs:
9733 * bindings/python/rtspserver.override:
9734 * bindings/python/rtspservermodule.c:
9735 * bindings/python/test.py:
9737 python: remove pygst-based python bindings
9738 pygi is the future, apparently.
9740 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
9743 Automatic update of common submodule
9744 From c463bc0 to 7fda524
9746 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9749 Automatic update of common submodule
9750 From 2a59016 to c463bc0
9752 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9755 Automatic update of common submodule
9756 From 0807187 to 2a59016
9758 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9761 Automatic update of common submodule
9762 From 11f0cd5 to 0807187
9764 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9766 * examples/test-auth.c:
9767 example: update for new caps
9769 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9771 * examples/test-video.c:
9772 * gst/rtsp-server/rtsp-client.c:
9773 * gst/rtsp-server/rtsp-media-factory-uri.c:
9774 * gst/rtsp-server/rtsp-media.c:
9775 * gst/rtsp-server/rtsp-media.h:
9776 * gst/rtsp-server/rtsp-session.c:
9777 * gst/rtsp-server/rtsp-session.h:
9778 rtsp-server: port some more to 0.11
9780 Remove bufferlist stuff
9782 Add queue before appsink now that preroll-queue-len is gone.
9783 Update for request pad changes.
9785 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9787 Merge branch 'master' into 0.11
9789 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9791 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9792 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9793 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9795 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9797 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9798 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9799 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9801 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9803 Merge branch 'master' into 0.11
9805 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9807 * gst/rtsp-server/rtsp-media.c:
9808 * gst/rtsp-server/rtsp-media.h:
9809 media: add a seekable boolean
9810 Maintain the seekable state with a new variable instead of reusing the
9813 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
9815 * gst/rtsp-server/rtsp-media.c:
9816 Disallow seek in live media
9818 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9820 Merge branch 'master' into 0.11
9822 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
9824 * gst/rtsp-server/rtsp-server.c:
9825 #ifdef statements for windows socket creation were missing
9827 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
9830 Automatic update of common submodule
9831 From a39eb83 to 11f0cd5
9833 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
9836 Automatic update of common submodule
9837 From 605cd9a to a39eb83
9839 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9841 Merge branch 'master' into 0.11
9843 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9845 * gst/rtsp-server/rtsp-client.c:
9846 client: use method to access property
9848 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9850 * gst/rtsp-server/rtsp-media-factory.c:
9851 * gst/rtsp-server/rtsp-media-factory.h:
9852 media-factory: add protocols property
9853 Add a property to configure the allowed protocols in the media created from the
9856 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9858 * gst/rtsp-server/rtsp-media-factory.c:
9859 * gst/rtsp-server/rtsp-media-factory.h:
9860 media-factory: add media-configure signal
9861 Add signal to allow the application to configure the media after it was created
9864 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9866 * gst/rtsp-server/rtsp-client.c:
9867 client: use method to access property
9869 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9871 * gst/rtsp-server/rtsp-media-factory.c:
9872 * gst/rtsp-server/rtsp-media-factory.h:
9873 media-factory: add protocols property
9874 Add a property to configure the allowed protocols in the media created from the
9877 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9879 * gst/rtsp-server/rtsp-media-factory.c:
9880 * gst/rtsp-server/rtsp-media-factory.h:
9881 media-factory: add media-configure signal
9882 Add signal to allow the application to configure the media after it was created
9885 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9887 Merge branch 'master' into 0.11
9889 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9891 * gst/rtsp-server/rtsp-client.c:
9892 client: use media multicast group
9894 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9896 * gst/rtsp-server/rtsp-media-factory.h:
9897 * gst/rtsp-server/rtsp-server.h:
9898 * gst/rtsp-server/rtsp-session-pool.h:
9899 * gst/rtsp-server/rtsp-session.h:
9902 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9904 * gst/rtsp-server/rtsp-client.c:
9905 * gst/rtsp-server/rtsp-sdp.h:
9906 sdp: copy and free the server ip address
9907 Copy and free the server ip address to make memory management easier later.
9909 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9911 * gst/rtsp-server/rtsp-media-factory.c:
9912 media-factory: configure multicast in media
9914 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9916 * gst/rtsp-server/rtsp-media.c:
9917 * gst/rtsp-server/rtsp-media.h:
9918 media: add property for multicast group
9919 Add a property to configure the multicast group in the media.
9920 Based on patches from Marc Leeman and Robert Krakora.
9922 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9924 * gst/rtsp-server/rtsp-media-factory.c:
9925 * gst/rtsp-server/rtsp-media-factory.h:
9926 media-factory: add property for multicast group
9927 Add a property to configure the multicast group in the media factory.
9928 Based on patches from Marc Leeman and Robert Krakora.
9930 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9932 * gst/rtsp-server/rtsp-client.c:
9933 client: do configuration of transport in one place
9934 Move the configuration of the transport destination address to where we also
9935 configure the other bits.
9937 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9939 * gst/rtsp-server/rtsp-client.c:
9940 client: use media multicast group
9942 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9944 * gst/rtsp-server/rtsp-media-factory.h:
9945 * gst/rtsp-server/rtsp-server.h:
9946 * gst/rtsp-server/rtsp-session-pool.h:
9947 * gst/rtsp-server/rtsp-session.h:
9950 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9952 * gst/rtsp-server/rtsp-client.c:
9953 * gst/rtsp-server/rtsp-sdp.h:
9954 sdp: copy and free the server ip address
9955 Copy and free the server ip address to make memory management easier later.
9957 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9959 * gst/rtsp-server/rtsp-media-factory.c:
9960 media-factory: configure multicast in media
9962 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9964 * gst/rtsp-server/rtsp-media.c:
9965 * gst/rtsp-server/rtsp-media.h:
9966 media: add property for multicast group
9967 Add a property to configure the multicast group in the media.
9968 Based on patches from Marc Leeman and Robert Krakora.
9970 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9972 * gst/rtsp-server/rtsp-media-factory.c:
9973 * gst/rtsp-server/rtsp-media-factory.h:
9974 media-factory: add property for multicast group
9975 Add a property to configure the multicast group in the media factory.
9976 Based on patches from Marc Leeman and Robert Krakora.
9978 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9980 * gst/rtsp-server/rtsp-client.c:
9981 client: do configuration of transport in one place
9982 Move the configuration of the transport destination address to where we also
9983 configure the other bits.
9985 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9987 Merge branch 'master' into 0.11
9989 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9991 * gst/rtsp-server/rtsp-client.c:
9992 client: destroy pipeline on client disconnect with no prior TEARDOWN.
9993 The problem occurs when the client abruptly closes the connection without
9994 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
9995 server is where the pipeline gets torn down. Since this handler is not called,
9996 the pipeline remains and is up and running. Subsequent clients get their own
9997 pipelines and if the do not issue TEARDOWNs then those pipelines will also
9998 remain up and running. This is a resource leak.
10000 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10002 Merge branch 'master' into 0.11
10004 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
10006 * gst/rtsp-server/rtsp-media-factory.c:
10007 * gst/rtsp-server/rtsp-media-factory.h:
10008 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
10009 For example, it can be used to retrieve source elements like appsrc, in a more
10010 convenient way than subclassing get_element.
10012 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10014 Merge branch 'master' into 0.11
10016 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
10018 * gst/rtsp-server/rtsp-server.c:
10019 rtsp-server: hold on to reference while using object
10021 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10023 * gst/rtsp-server/rtsp-media.c:
10026 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10029 configure: use unstable api
10031 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
10033 * gst/rtsp-server/rtsp-client.c:
10034 client: fix reference counting
10036 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
10038 * gst/rtsp-server/rtsp-client.c:
10039 * gst/rtsp-server/rtsp-media.c:
10040 fix compiler warnings about unused variables
10042 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
10044 * examples/test-launch.c:
10045 * examples/test-readme.c:
10046 * examples/test-uri.c:
10047 * examples/test-video.c:
10048 examples: tell rtsp uri when ready
10050 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
10053 Automatic update of common submodule
10054 From 69b981f to 605cd9a
10056 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10058 * gst/rtsp-server/rtsp-client.c:
10059 client: update for buffer API change
10061 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10063 * gst/rtsp-server/Makefile.am:
10064 Makefile.am: 0.10 => @GST_MAJORMINOR@
10066 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10068 * gst/rtsp-server/rtsp-media-factory-uri.c:
10069 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
10071 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10073 * gst/rtsp-server/.gitignore:
10074 .gitignore: 0.10 => 0.11
10076 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10078 * gst/rtsp-server/Makefile.am:
10079 Makefile.am: 0.10 => @GST_MAJORMINOR@
10081 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10083 Merge branch 'master' into 0.11
10085 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
10088 Automatic update of common submodule
10089 From 9e5bbd5 to 69b981f
10091 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
10094 Automatic update of common submodule
10095 From fd35073 to 9e5bbd5
10097 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
10100 Automatic update of common submodule
10101 From 46dfcea to fd35073
10103 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10105 * gst/rtsp-server/rtsp-media-factory-uri.c:
10106 * gst/rtsp-server/rtsp-media.c:
10107 media: port to new caps API
10109 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10111 Merge branch 'master' into 0.11
10113 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10115 * bindings/vala/gst-rtsp-server-0.10.vapi:
10116 Updated Vala bindings.
10117 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10119 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10121 * gst/rtsp-server/rtsp-server.c:
10122 * gst/rtsp-server/rtsp-server.h:
10123 Add a signal for newly connected clients.
10124 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10126 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
10128 * bindings/python/rtspserver.override:
10129 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
10131 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10133 * gst/rtsp-server/Makefile.am:
10134 * gst/rtsp-server/rtsp-client.c:
10135 * gst/rtsp-server/rtsp-funnel.c:
10136 * gst/rtsp-server/rtsp-funnel.h:
10137 * gst/rtsp-server/rtsp-media.c:
10138 rtsp-server: port to 0.11
10140 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10145 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10147 Merge branch 'master' into 0.11
10152 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10155 Automatic update of common submodule
10156 From c3cafe1 to 46dfcea
10158 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
10160 * bindings/python/Makefile.am:
10161 * bindings/python/rtspserver.defs:
10162 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
10164 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
10166 * bindings/python/arg-types.py:
10167 python bindings: add GstRTSPUrlParam
10168 Needed to implement MediaFactory virtual proxies
10170 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
10172 * bindings/python/arg-types.py:
10173 python bindings: fix returning GstRTSPUrl types
10175 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
10177 * bindings/python/arg-types.py:
10178 python bindings: add arg type for GstRTSPUrl
10180 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
10182 * bindings/python/rtspserver.defs:
10183 python bindings: fix the definition of MediaFactory.collect_stream
10185 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
10188 Automatic update of common submodule
10189 From 1ccbe09 to c3cafe1
10191 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10194 Automatic update of common submodule
10195 From 193b717 to 1ccbe09
10197 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
10200 Automatic update of common submodule
10201 From b77e2bf to 193b717
10203 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10206 build: Include lcov.mak to allow test coverage report generation
10208 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10211 Automatic update of common submodule
10212 From d8814b6 to b77e2bf
10214 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10217 Automatic update of common submodule
10218 From 6aaa286 to d8814b6
10220 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
10223 Automatic update of common submodule
10224 From 6aec6b9 to 6aaa286
10226 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
10229 autogen: wingo signed comment
10231 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
10233 * gst/rtsp-server/rtsp-session-pool.c:
10234 session: use full charset for RTSP session ID
10235 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
10236 session ID more difficult.
10237 https://bugzilla.gnome.org/show_bug.cgi?id=643812
10239 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10241 * gst/rtsp-server/Makefile.am:
10242 rtsp-server: Don't install the funnel header
10244 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10247 Automatic update of common submodule
10248 From 1de7f6a to 6aec6b9
10250 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10253 configure: require core/base 0.10.31
10254 Needed at least for gst_plugin_feature_rank_compare_func().
10256 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
10259 Automatic update of common submodule
10260 From f94d739 to 1de7f6a
10262 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10264 * gst/rtsp-server/rtsp-media.c:
10265 media: remove more unused code
10267 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10269 * gst/rtsp-server/rtsp-media.c:
10270 * gst/rtsp-server/rtsp-media.h:
10271 media: remove duplicate filtering
10272 Remove the duplicate filtering code now that we have a released -good version.
10273 Give a warning instead.
10275 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10277 * gst/rtsp-server/rtsp-media-factory.c:
10278 * gst/rtsp-server/rtsp-media.c:
10279 media: fix default buffer size
10281 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10283 * gst/rtsp-server/rtsp-media-factory.c:
10284 * gst/rtsp-server/rtsp-media-factory.h:
10285 media-factory: add property to configure the buffer-size
10286 Add a property to configure the kernel UDP buffer size.
10288 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10290 * gst/rtsp-server/rtsp-media.c:
10291 * gst/rtsp-server/rtsp-media.h:
10292 media: add property to configure kernel buffer sizes
10293 Add a property to configure the kernel UDP buffer size.
10295 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10298 configure: set PYGOBJECT_REQ before using it
10299 https://bugzilla.gnome.org/show_bug.cgi?id=640641
10301 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10303 * docs/Makefile.am:
10304 docs: recursive into sub-directories on 'make upload'
10306 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10308 * docs/libs/gst-rtsp-server-docs.sgml:
10309 * docs/version.entities.in:
10310 docs: mention full version these docs are for, not just major-minor
10312 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10315 back to development
10317 === release 0.10.8 ===
10319 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10324 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10326 * gst/rtsp-server/rtsp-server.c:
10327 rtsp-server: clarify docs a little
10329 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10331 * gst/rtsp-server/rtsp-media.c:
10332 media: init debug category before starting thread
10334 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10336 * gst/rtsp-server/rtsp-auth.c:
10337 auth: add realm to make it more spec compliant
10339 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10341 * gst/rtsp-server/rtsp-server.c:
10342 * gst/rtsp-server/rtsp-server.h:
10343 server: add locking
10345 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10347 * examples/test-video.c:
10348 example: improve example docs a little
10350 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10352 * gst/rtsp-server/rtsp-server.c:
10353 server: ensure the watch has a ref to the server
10355 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10357 * gst/rtsp-server/rtsp-server.c:
10358 server: simpify channel function
10360 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10362 * gst/rtsp-server/rtsp-server.c:
10363 * gst/rtsp-server/rtsp-server.h:
10364 server: simplify management of channel and source
10365 We don't need to keep around the channel and source objects. Let the mainloop
10366 and the source manage the source and channel respectively.
10368 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10374 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10376 * tests/.gitignore:
10377 * tests/Makefile.am:
10378 * tests/test-cleanup.c:
10379 tests: add tests directory and cleanup test
10381 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10383 * gst/rtsp-server/rtsp-media-factory-uri.c:
10384 * gst/rtsp-server/rtsp-media-factory.c:
10385 * gst/rtsp-server/rtsp-media-mapping.c:
10386 * gst/rtsp-server/rtsp-media.c:
10387 * gst/rtsp-server/rtsp-session-pool.c:
10388 * gst/rtsp-server/rtsp-session.c:
10389 server: improve debugging in various objects
10391 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10393 * gst/rtsp-server/rtsp-server.c:
10394 server: chain up to the parent finalize
10396 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
10398 * bindings/python/rtspserver-types.defs:
10399 * bindings/python/rtspserver.defs:
10400 * bindings/python/rtspserver.override:
10401 * bindings/python/test.py:
10402 gst-rtsp-server: update python bindings
10404 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10406 * gst/rtsp-server/rtsp-client.c:
10407 client: use the response from the clientstate
10408 Create the response object only once and store in the client state.
10409 Make all methods use the state response,
10411 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10413 * gst/rtsp-server/rtsp-server.c:
10414 server: use signal to keep track of clients
10415 Keep track of all the clients that the server creates and remove them when they
10416 fire the 'closed' signal.
10418 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10420 * gst/rtsp-server/rtsp-client.c:
10421 * gst/rtsp-server/rtsp-client.h:
10422 client: emit signal when closing
10424 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10426 * examples/.gitignore:
10427 * examples/Makefile.am:
10428 * examples/test-auth.c:
10429 * examples/test-video.c:
10430 * gst/rtsp-server/rtsp-auth.c:
10431 * gst/rtsp-server/rtsp-auth.h:
10432 * gst/rtsp-server/rtsp-client.c:
10433 * gst/rtsp-server/rtsp-media-factory.c:
10434 * gst/rtsp-server/rtsp-media.c:
10435 * gst/rtsp-server/rtsp-media.h:
10436 * gst/rtsp-server/rtsp-session-pool.h:
10437 * gst/rtsp-server/rtsp-session.h:
10438 media: enable per factory authorisations
10439 Allow for adding a GstRTSPAuth on the factory and media level and check
10440 permissions when accessing the factory.
10441 Add hints to the auth methods for future more fine grained authorisation.
10442 Add example application for per factory authentication.
10444 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10446 * gst/rtsp-server/rtsp-auth.c:
10447 * gst/rtsp-server/rtsp-auth.h:
10448 * gst/rtsp-server/rtsp-client.c:
10449 * gst/rtsp-server/rtsp-client.h:
10450 * gst/rtsp-server/rtsp-params.c:
10451 * gst/rtsp-server/rtsp-params.h:
10452 rtsp-server: Pass ClientState structure arround
10453 Pass the collected information for the ongoing request in a GstRTSPClientState
10454 structure that we can then pass around to simplify the method arguments. This
10455 will also be handy when we implement logging functionality.
10457 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10459 * gst/rtsp-server/rtsp-media-factory.c:
10460 * gst/rtsp-server/rtsp-media-factory.h:
10461 media-factory: add methods to configure authorisation
10463 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10465 * gst/rtsp-server/rtsp-client.c:
10466 client: unref auth in finalize
10468 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10470 * gst/rtsp-server/rtsp-server.c:
10471 server: unref auth in finalize
10473 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10475 * docs/libs/gst-rtsp-server-docs.sgml:
10476 * docs/libs/gst-rtsp-server-sections.txt:
10477 * docs/libs/gst-rtsp-server.types:
10478 docs: add more docs
10480 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10482 * gst/rtsp-server/rtsp-server.c:
10483 * gst/rtsp-server/rtsp-server.h:
10484 server: separate create and accept
10485 Create separate create and accept methods so that subclasses can create custom
10487 Configure the server in the client object and prepare for keeping track of
10490 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10492 * gst/rtsp-server/rtsp-client.c:
10493 * gst/rtsp-server/rtsp-client.h:
10494 client: add support for setting the server.
10495 Add support for keeping a ref to the server that started this client
10498 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10500 * gst/rtsp-server/rtsp-auth.c:
10501 auth: fix memleak and add some docs
10502 Fix a memleak of the basic auth token.
10503 Add docs for the helper function
10505 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10507 * gst/rtsp-server/rtsp-auth.c:
10508 * gst/rtsp-server/rtsp-auth.h:
10509 * gst/rtsp-server/rtsp-client.c:
10510 client: delegate setup of auth to the manager
10511 Delegate the configuration of the authentication tokens to the manager object
10514 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10516 * examples/test-video.c:
10517 * gst/rtsp-server/Makefile.am:
10518 * gst/rtsp-server/rtsp-auth.c:
10519 * gst/rtsp-server/rtsp-auth.h:
10520 * gst/rtsp-server/rtsp-client.c:
10521 * gst/rtsp-server/rtsp-client.h:
10522 * gst/rtsp-server/rtsp-server.c:
10523 * gst/rtsp-server/rtsp-server.h:
10524 auth: add authentication object
10525 Add an object that can check the authorization of requests.
10526 Implement basic authentication.
10527 Add example authentication to test-video
10529 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10531 * gst/rtsp-server/rtsp-server.c:
10532 * gst/rtsp-server/rtsp-server.h:
10533 server: move includes back
10534 the includes are needed for sockaddr_in.
10536 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10538 * gst/rtsp-server/rtsp-client.c:
10539 * gst/rtsp-server/rtsp-client.h:
10540 * gst/rtsp-server/rtsp-server.c:
10541 * gst/rtsp-server/rtsp-server.h:
10542 rtsp: move network includes where they are needed
10544 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
10546 * gst/rtsp-server/rtsp-media.h:
10547 rtsp-media.h: Minor corrections in comments.
10550 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
10553 Automatic update of common submodule
10554 From e572c87 to f94d739
10556 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10560 * docs/libs/.gitignore:
10561 * examples/.gitignore:
10562 * gst/rtsp-server/.gitignore:
10565 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10567 * docs/libs/Makefile.am:
10568 docs: We don't build ps/pdf for API reference docs
10570 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10573 Automatic update of common submodule
10574 From ccbaa85 to e572c87
10576 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10579 Automatic update of common submodule
10580 From 46445ad to ccbaa85
10582 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10584 * gst/rtsp-server/Makefile.am:
10585 * gst/rtsp-server/rtsp-funnel.c:
10586 * gst/rtsp-server/rtsp-funnel.h:
10587 * gst/rtsp-server/rtsp-media.c:
10588 funnel: rename fsfunnel to rtspfunnel
10589 Rename the funnel to avoid conflicts with the farsight one.
10591 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10593 * gst/rtsp-server/Makefile.am:
10594 * gst/rtsp-server/fs-funnel.c:
10595 * gst/rtsp-server/fs-funnel.h:
10596 * gst/rtsp-server/rtsp-media.c:
10597 rtsp-media: add and use fsfunnel
10598 Add a copy of fsfunnel to the build because input-selector removed the (broken)
10599 select-all property that we need.
10601 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10603 * gst/rtsp-server/Makefile.am:
10604 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
10605 Use PKG_CONFIG_PATH specified at configure time (if any) as well
10606 for the g-ir-compiler, rather than just assuming the env var has
10609 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10616 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
10618 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10621 * gst/rtsp-server/Makefile.am:
10622 gobject-introspection: fix g-i build for uninstalled setup
10623 Requires gst-plugins-base git (> 0.10.31.2).
10625 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10627 * examples/test-uri.c:
10628 examples: add some more options and comments
10630 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10632 * gst/rtsp-server/rtsp-media-factory-uri.c:
10633 factory-uri: use right property type
10635 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10637 * gst/rtsp-server/rtsp-media-factory-uri.c:
10638 factory-uri: attempt to configure buffer-lists
10639 Attempt to configure buffer lists in the payloader for improved performance.
10641 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10643 * gst/rtsp-server/rtsp-media.c:
10644 media: attempt to configure bigger UDP buffers
10645 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
10646 send buffers with high bitrate streams.
10648 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
10650 * gst/rtsp-server/rtsp-client.c:
10651 client: use the socket length from getsockname
10652 Use the length returned by getsockname to perform the getnameinfo call because
10653 the size can depend on the socket type and platform.
10656 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10658 * docs/libs/gst-rtsp-server-docs.sgml:
10659 * docs/libs/gst-rtsp-server-sections.txt:
10660 docs: add uri factory to the docs
10662 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10664 * gst/rtsp-server/rtsp-client.c:
10665 * gst/rtsp-server/rtsp-media.h:
10668 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10670 * gst/rtsp-server/rtsp-client.c:
10671 * gst/rtsp-server/rtsp-media.c:
10672 * gst/rtsp-server/rtsp-media.h:
10673 * gst/rtsp-server/rtsp-session.c:
10674 * gst/rtsp-server/rtsp-session.h:
10675 rtsp-server: add support for buffer lists
10676 Add support for sending bufferlists received from appsink.
10679 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10681 * gst/rtsp-server/rtsp-client.c:
10682 * gst/rtsp-server/rtsp-media.c:
10683 * gst/rtsp-server/rtsp-media.h:
10684 * gst/rtsp-server/rtsp-sdp.c:
10685 media: make method to retrieve the play range
10686 Make a method to retrieve the playback range so that we can conditionally create
10687 a different range for the SDP and the PLAY requests.
10689 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10691 * gst/rtsp-server/rtsp-media.c:
10692 * gst/rtsp-server/rtsp-media.h:
10693 media: add signal to notify of state changes
10695 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10697 * gst/rtsp-server/rtsp-client.h:
10698 client: cleanup headers
10700 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10702 * gst/rtsp-server/rtsp-client.c:
10705 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10707 * gst/rtsp-server/rtsp-media-factory-uri.c:
10708 * gst/rtsp-server/rtsp-media-factory-uri.h:
10709 factory-uri: add support for gstpay
10710 Add an option to prefer gstpay over decoder + raw payloader.
10712 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10714 * gst/rtsp-server/rtsp-media-factory-uri.c:
10715 * gst/rtsp-server/rtsp-media-factory-uri.h:
10716 factory-uri: rework the autoplugger.
10717 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
10720 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10722 * gst/rtsp-server/rtsp-media-factory-uri.c:
10723 factory-uri: use better factory filter
10724 Make better payloader filter based on autoplug rank and RTP use case.
10726 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10729 Automatic update of common submodule
10730 From 169462a to 46445ad
10732 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10734 * gst/rtsp-server/rtsp-server.c:
10735 server: set SO_REUSEADDR before bind
10736 Set the SO_REUSEADDR _before_ bind() to make it actually work.
10738 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10740 * gst/rtsp-server/rtsp-media.c:
10741 * gst/rtsp-server/rtsp-media.h:
10742 media: emit prepared signal when prepared
10743 Make a 'prepared' signal and emit it when we successfully prepared the element.
10744 This signal can be used to configure the media object after it has been prepared
10747 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
10750 Automatic update of common submodule
10751 From 011bcc8 to 169462a
10753 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
10755 python an optional dependency
10756 * configure.ac: Move up valgrind and g-i checks. Make the python
10757 dependency optional, as it was before.
10759 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10761 Merge branch 'master' into 0.11
10766 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10768 * gst/rtsp-server/rtsp-media.c:
10769 media: update range when active clients changed
10770 When we changed the number of active clients, update the current range
10771 information because we want the second client connecting to a shared resource
10772 continue from where the stream currently.
10774 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10776 * gst/rtsp-server/rtsp-media-factory-uri.c:
10777 * gst/rtsp-server/rtsp-media-factory-uri.h:
10778 factory-uri: add colorspace and fix pt
10779 Rework the way we pass data to the autoplugger.
10780 When we have raw caps, plug a converter element to make pluggin to raw
10781 payloaders more successful.
10782 Make sure all dynamically plugged payloaders have a unique payload types.
10784 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10786 * examples/Makefile.am:
10787 * examples/test-uri.c:
10788 example: add example of the uri factory
10790 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10792 * gst/rtsp-server/Makefile.am:
10793 * gst/rtsp-server/rtsp-media-factory-uri.c:
10794 * gst/rtsp-server/rtsp-media-factory-uri.h:
10795 * gst/rtsp-server/rtsp-server.h:
10796 factory-uri: add a factory to stream any URI
10797 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
10800 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-media.c:
10803 * gst/rtsp-server/rtsp-media.h:
10804 media: ignore spurious ASYNC_DONE messages
10805 When we are dynamically adding pads, the addition of the udpsrc elements will
10806 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
10807 the real ASYNC_DONE when everything is prerolled.
10809 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10811 * gst/rtsp-server/rtsp-media-factory.c:
10812 * gst/rtsp-server/rtsp-media-factory.h:
10813 media-factory: make lock macro
10815 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
10817 * gst/rtsp-server/rtsp-client.c:
10818 rtsp-server: Remove unused variable and dead assignment
10820 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
10822 * examples/test-launch.c:
10823 * examples/test-mp4.c:
10824 * examples/test-ogg.c:
10825 * examples/test-readme.c:
10826 * examples/test-sdp.c:
10827 * examples/test-video.c:
10828 examples: Run gst-indent
10830 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
10832 * gst/rtsp-server/rtsp-client.c:
10833 * gst/rtsp-server/rtsp-media-factory.c:
10834 * gst/rtsp-server/rtsp-media-mapping.c:
10835 * gst/rtsp-server/rtsp-media.c:
10836 * gst/rtsp-server/rtsp-params.c:
10837 * gst/rtsp-server/rtsp-sdp.c:
10838 * gst/rtsp-server/rtsp-server.c:
10839 * gst/rtsp-server/rtsp-session-pool.c:
10840 * gst/rtsp-server/rtsp-session.c:
10841 rtsp-server: Run gst-indent
10842 Since it wasn't using the upstream common previously, there was no
10843 indentation check before commiting.
10845 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
10847 * gst/rtsp-server/rtsp-media-mapping.h:
10848 * gst/rtsp-server/rtsp-media.c:
10849 * gst/rtsp-server/rtsp-media.h:
10850 * gst/rtsp-server/rtsp-sdp.c:
10851 * gst/rtsp-server/rtsp-session-pool.h:
10852 * gst/rtsp-server/rtsp-session.c:
10853 * gst/rtsp-server/rtsp-session.h:
10854 rtsp-server: Some more doc fixups
10856 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10859 Makefile: Add cruft-cleaning support
10861 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10865 * docs/Makefile.am:
10866 * docs/libs/Makefile.am:
10867 * docs/libs/gst-rtsp-server-docs.sgml:
10868 * docs/libs/gst-rtsp-server-sections.txt:
10869 * docs/libs/gst-rtsp-server.types:
10870 * docs/version.entities.in:
10871 docs: Add gtk-doc build system
10873 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10875 * gst/rtsp-server/Makefile.am:
10876 Makefile.am: Use standard GIR make behaviour
10878 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10882 autogen/configure: Bring more in sync to standard gst module behaviour
10884 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10886 * gst/rtsp-server/rtsp-media.c:
10887 media: warn and fail when gstrtpbin is not found
10889 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10892 configure: open 0.11 branch
10894 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
10898 Add common submodule
10900 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
10902 * common/ChangeLog:
10903 * common/Makefile.am:
10904 * common/c-to-xml.py:
10905 * common/check.mak:
10906 * common/coverage/coverage-report-entry.pl:
10907 * common/coverage/coverage-report.pl:
10908 * common/coverage/coverage-report.xsl:
10909 * common/coverage/lcov.mak:
10910 * common/gettext.patch:
10911 * common/glib-gen.mak:
10912 * common/gst-autogen.sh:
10913 * common/gst-xmlinspect.py:
10915 * common/gstdoc-scangobj:
10916 * common/gtk-doc-plugins.mak:
10917 * common/gtk-doc.mak:
10918 * common/m4/.gitignore:
10919 * common/m4/Makefile.am:
10920 * common/m4/README:
10921 * common/m4/as-ac-expand.m4:
10922 * common/m4/as-auto-alt.m4:
10923 * common/m4/as-compiler-flag.m4:
10924 * common/m4/as-compiler.m4:
10925 * common/m4/as-docbook.m4:
10926 * common/m4/as-libtool-tags.m4:
10927 * common/m4/as-libtool.m4:
10928 * common/m4/as-python.m4:
10929 * common/m4/as-scrub-include.m4:
10930 * common/m4/as-version.m4:
10931 * common/m4/ax_create_stdint_h.m4:
10932 * common/m4/check.m4:
10933 * common/m4/glib-gettext.m4:
10934 * common/m4/gst-arch.m4:
10935 * common/m4/gst-args.m4:
10936 * common/m4/gst-check.m4:
10937 * common/m4/gst-debuginfo.m4:
10938 * common/m4/gst-default.m4:
10939 * common/m4/gst-doc.m4:
10940 * common/m4/gst-error.m4:
10941 * common/m4/gst-feature.m4:
10942 * common/m4/gst-function.m4:
10943 * common/m4/gst-gettext.m4:
10944 * common/m4/gst-glib2.m4:
10945 * common/m4/gst-libxml2.m4:
10946 * common/m4/gst-plugindir.m4:
10947 * common/m4/gst-valgrind.m4:
10948 * common/m4/gtk-doc.m4:
10949 * common/m4/introspection.m4:
10950 * common/m4/pkg.m4:
10951 * common/mangle-tmpl.py:
10952 * common/plugins.xsl:
10954 * common/release.mak:
10955 * common/scangobj-merge.py:
10956 * common/upload.mak:
10957 common: Remove static version
10959 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
10961 * common/m4/introspection.m4:
10962 Update introspection.m4 to match usage
10964 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10968 Remove old stuff from the README
10970 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10973 back to development
10975 === release 0.10.7 ===
10977 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10982 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10984 * examples/test-ogg.c:
10985 test-ogg: remove parsers
10986 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
10987 buffers with timestamps. Using the parsers also seems to break things.
10989 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10991 * bindings/vala/gst-rtsp-server-0.10.vapi:
10992 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10993 Updated Vala bindings
10995 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10997 * common/m4/introspection.m4:
10999 * gst/rtsp-server/Makefile.am:
11000 Added initial gobject-introspection support
11002 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11004 * gst/rtsp-server/rtsp-media-factory.c:
11005 media-factory: don't use host for shared hash key
11006 When we generate the key to share made between connections, don't include the
11007 host used to connect so that we can share media even if between clients that
11008 connected with localhost and ones with the ip address.
11010 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11012 * bindings/vala/Makefile.am:
11013 build: fix distcheck
11015 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11017 * bindings/vala/gst-rtsp-server-0.10.vapi:
11018 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11019 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11020 Update Vala bindings
11022 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11024 * bindings/vala/Makefile.am:
11026 Fix configure checks and installation location for Vala bindings
11029 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11032 back to development
11034 === release 0.10.6 ===
11036 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11039 configure: release 0.10.6
11041 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11043 * gst/rtsp-server/rtsp-media.c:
11044 media: help the compiler a little
11046 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11048 * gst/rtsp-server/rtsp-media.c:
11049 * gst/rtsp-server/rtsp-media.h:
11050 * gst/rtsp-server/rtsp-session.c:
11051 media: cleanup media transport before freeing
11052 Cleanup the media transport data before freeing. In particular, remove the qdata
11053 from the rtpsource object.
11055 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11057 * gst/rtsp-server/rtsp-media-factory.c:
11058 * gst/rtsp-server/rtsp-media-factory.h:
11059 * gst/rtsp-server/rtsp-media.c:
11060 * gst/rtsp-server/rtsp-media.h:
11061 media-factory: add eos-shutdown property
11062 Add an eos-shutdown property that will send an EOS to the pipeline before
11063 shutting it down. This allows for nice cleanup in case of a muxer.
11066 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11068 * gst/rtsp-server/rtsp-media.c:
11069 * gst/rtsp-server/rtsp-media.h:
11070 media: use multiudpsink send-duplicates when we can
11071 If we have a new enough multiudpsink with the send-duplicates property, use this
11072 instead of doing our own filtering. Our custom filtering code should eventually
11073 be removed when we can depend on a released -good.
11075 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11077 * gst/rtsp-server/rtsp-media.c:
11078 media: don't leak destinations
11079 Refactor and cleanup the destinations array when the stream is destroyed.
11081 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11083 * gst/rtsp-server/rtsp-media.c:
11084 * gst/rtsp-server/rtsp-media.h:
11085 media: don't add udp addresses multiple times
11086 Keep track of the udp addresses we added to udpsink and never add the same udp
11087 destination twice. This avoids duplicate packets when using multicast.
11089 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11091 * gst/rtsp-server/rtsp-server.c:
11092 server: disable use of SO_LINGER
11093 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
11094 server close()s the connection.
11096 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11098 * gst/rtsp-server/rtsp-server.c:
11099 server: use 5 second linger period in SO_LINGER
11100 Wait 5 seconds before clearing the send buffers and reseting the connection with
11101 the client when we do a close. This should be enough time to get the message to
11105 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11107 * gst/rtsp-server/rtsp-server.c:
11108 server: use SO_LINGER
11109 SO_LINGER on the socket will make sure that any pending data on the socket is
11110 flushed ASAP and that the socket connection is reset. This makes sure that the
11111 socket can be reused immediately.
11114 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11117 README: add blurb about shared media factories
11119 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
11121 * gst/rtsp-server/rtsp-media.c:
11122 Add stdlib.h for atoi()
11124 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11126 * bindings/python/Makefile.am:
11127 * bindings/vala/Makefile.am:
11128 build: distcheck fixes
11129 Fix 'make distcheck', somewhat (it still fails because it tries to
11130 install files into /usr/share/vala/vapi/ irrespective of the
11131 configured prefix).
11133 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11136 configure: bump core/base requirements to released version
11137 Makes things less confusing for people.
11139 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11142 configure: fail if GStreamer core/base requirements are not met
11144 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11146 * gst/rtsp-server/rtsp-client.c:
11147 client: improve client cleanups
11148 Make sure the session does not timeout when using TCP. We need to do this
11149 because quicktime player does not send RTCP for some reason in tunneled
11151 Refactor some cleanup code.
11154 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11156 * gst/rtsp-server/rtsp-session.c:
11157 * gst/rtsp-server/rtsp-session.h:
11158 session: add support for prevent session timeouts
11159 Add an atomix counter to prevent session timeouts when we are, for example,
11160 streaming over TCP.
11162 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11164 * gst/rtsp-server/rtsp-client.c:
11165 client: fix unlink on session timeouts
11166 When our session times out, make sure we unlink all streams in this
11168 Remove the tunnelid when closing the connection.
11170 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11172 * gst/rtsp-server/rtsp-session.c:
11173 session: small cleanups
11175 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11177 * gst/rtsp-server/rtsp-client.c:
11178 client: handle lost_tunnel callbacks
11179 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
11180 hashtable so that we can reuse it for when the client reopens the POST
11182 Close the connection after a TEARDOWN.
11183 Make sure or watchid is cleared when the watch is removed.
11186 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11188 * gst/rtsp-server/rtsp-client.c:
11189 * gst/rtsp-server/rtsp-media.c:
11190 * gst/rtsp-server/rtsp-sdp.c:
11191 rtsp-server: add more support for multicast
11193 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11196 * gst/rtsp-server/rtsp-media.c:
11197 * gst/rtsp-server/rtsp-media.h:
11198 media: allow configuration of allowed lower transport
11200 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11202 * gst/rtsp-server/rtsp-client.h:
11203 * gst/rtsp-server/rtsp-media.c:
11204 * gst/rtsp-server/rtsp-media.h:
11205 * gst/rtsp-server/rtsp-sdp.c:
11206 * gst/rtsp-server/rtsp-sdp.h:
11207 * gst/rtsp-server/rtsp-server.c:
11208 rtsp: keep track of server ip and ipv6
11209 Keep track of how the client connected to the server and setup the udp ports
11210 with the same protocol.
11211 Copy the server ip address in the SDP so that clients can send RTCP back to
11214 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11216 * gst/rtsp-server/rtsp-session.c:
11219 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11221 * gst/rtsp-server/rtsp-client.c:
11222 client: use right size for malloc
11224 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11226 * gst/rtsp-server/rtsp-server.c:
11227 server: comment ipv6 server listening address
11229 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11231 * gst/rtsp-server/rtsp-media.c:
11232 media: allow for ipv6 sockets
11234 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11236 * gst/rtsp-server/rtsp-server.c:
11237 * gst/rtsp-server/rtsp-server.h:
11238 server: rework server part
11239 Allow setting a bind address, make sure we can deal with ipv6.
11240 Remove the port property and change with the service property.
11242 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11244 * gst/rtsp-server/rtsp-media.h:
11245 media: update comments a little
11247 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11249 * gst/rtsp-server/rtsp-client.c:
11250 client: make content-base better
11251 Use the URI formatting functions to make a content-base. Also make sure that
11252 there is a trailing / at the end.
11254 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11256 * gst/rtsp-server/rtsp-client.c:
11257 client: guard against invalid paths
11259 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11261 * examples/test-video.c:
11262 test: catch server bind errors
11264 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
11266 * gst/rtsp-server/rtsp-media.c:
11267 rtspmedia: emit "unprepared" if _prepare fails.
11268 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
11269 media object is removed from its factory's cache.
11271 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11273 * gst/rtsp-server/rtsp-media.c:
11274 media: collect media position when seek completes
11276 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
11278 * gst/rtsp-server/rtsp-client.c:
11279 client: call unlink_streams in client finalize
11282 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11284 * gst/rtsp-server/rtsp-media.c:
11285 media: limit the time to wait to something huge
11286 Avoid waiting forever but limit the timeout to 20 seconds.
11288 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11290 * gst/rtsp-server/rtsp-sdp.c:
11291 sdp: reindent and check for prepared status
11293 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11295 * gst/rtsp-server/rtsp-media.c:
11296 * gst/rtsp-server/rtsp-media.h:
11297 * gst/rtsp-server/rtsp-session.c:
11298 media: avoid doing _get_state() for state changes
11299 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
11300 until the media is prerolled or in error. This avoids doing a blocking call of
11301 gst_element_get_state() that can cause lockups when there is an error.
11304 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11306 * gst/rtsp-server/rtsp-media.c:
11309 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11311 * gst/rtsp-server/rtsp-media-factory.c:
11312 media-factory: better error handling
11313 Improve the error handling a bit.
11315 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11317 * gst/rtsp-server/rtsp-client.c:
11318 client: rework transport parsing
11319 Rework the transport parsing code so that we can ignore transports we don't
11320 support instead of just picking the first one we can parse.
11321 Configure a (for now hardcoded) destination for multicast transports.
11323 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11325 * gst/rtsp-server/rtsp-media.c:
11326 media: set multicast sink parameters
11327 Disable loop and automatic multicast join on the udpsink elements.
11328 Add some more debug info.
11329 Reset some state variables in the right place.
11330 Use the right port numbers for multicast.
11332 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11334 * gst/rtsp-server/rtsp-session.c:
11335 session: handle transport setup correctly
11336 Handle UDP, MCAST and TCP transport negotiation more correctly.
11337 Store the server session SSRC in the transport.
11339 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11341 * gst/rtsp-server/rtsp-client.c:
11342 rtsp-client: implement error_full
11343 Implement error_full to avoid some segfaults when the rtspconnection calls it.
11346 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11349 * gst/rtsp-server/rtsp-client.c:
11350 * gst/rtsp-server/rtsp-server.c:
11351 docs: update docs and comments
11353 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
11355 * gst/rtsp-server/rtsp-sdp.c:
11356 sdp: make server work better when behind a proxy
11358 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11360 * gst/rtsp-server/rtsp-client.c:
11361 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
11363 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11365 * gst/rtsp-server/rtsp-client.c:
11366 * gst/rtsp-server/rtsp-media-factory.c:
11367 * gst/rtsp-server/rtsp-media-mapping.c:
11368 * gst/rtsp-server/rtsp-media.c:
11369 * gst/rtsp-server/rtsp-server.c:
11370 * gst/rtsp-server/rtsp-session-pool.c:
11371 * gst/rtsp-server/rtsp-session.c:
11372 Use GStreamer's debugging subsystem
11374 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11376 * gst/rtsp-server/rtsp-media-factory.c:
11377 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
11379 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11382 back to development
11384 === release 0.10.5 ===
11386 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11391 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11394 configure: bump required versions
11396 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
11398 * gst/rtsp-server/rtsp-client.c:
11399 client: call weak-unref on client->sessions from finalize
11402 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11404 * gst/rtsp-server/rtsp-media.c:
11405 media: Fixed crasher where caps got unref'ed too often
11407 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11410 * pkgconfig/.gitignore:
11411 * pkgconfig/Makefile.am:
11412 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
11413 Added pkg-config file to use gst-rtsp-server uninstalled
11415 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11417 * gst/rtsp-server/rtsp-media.c:
11418 media: add some docs
11420 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
11422 * gst/rtsp-server/rtsp-client.c:
11423 rtsp: Use gst_rtsp_watch_send_message().
11424 Use gst_rtsp_watch_send_message() since the old API which used
11425 gst_rtsp_watch_queue_message() has been deprecated.
11427 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11430 back to development
11432 === release 0.10.4 ===
11434 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11439 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11441 * gst/rtsp-server/rtsp-client.c:
11442 * gst/rtsp-server/rtsp-session.c:
11443 * gst/rtsp-server/rtsp-session.h:
11444 rtsp: allocate channels in TCP mode
11445 When the client does not provide us with channels in TCP mode, allocate channels
11448 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11450 * gst/rtsp-server/rtsp-client.c:
11451 client: don't crash when tunnelid is missing
11452 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
11453 don't crash but return an error response to the client.
11456 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11458 * bindings/vala/gst-rtsp-server-0.10.vapi:
11459 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11460 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11461 bindings: update vala bindings with new method
11463 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11465 * gst/rtsp-server/rtsp-session-pool.c:
11466 * gst/rtsp-server/rtsp-session-pool.h:
11467 sessionpool: add function to filter sessions
11468 Add generic function to retrieve/remove sessions.
11470 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11473 configure: bump core/base requirements to release
11475 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11477 * gst/rtsp-server/rtsp-media.c:
11478 media: fix indentation
11480 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11482 * gst/rtsp-server/rtsp-media.c:
11483 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
11485 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11487 * gst/rtsp-server/rtsp-media.c:
11488 set state and remove elements of media in for loop
11490 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
11492 * bindings/vala/gst-rtsp-server-0.10.vapi:
11493 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11494 Added gst_rtsp_media_remove_elements function to Vala bindings
11496 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
11498 * gst/rtsp-server/rtsp-media.c:
11499 * gst/rtsp-server/rtsp-media.h:
11500 Added gst_rtsp_media_remove_elements function
11502 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
11504 * gst/rtsp-server/rtsp-media.c:
11505 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
11507 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11509 * bindings/vala/gst-rtsp-server-0.10.vapi:
11510 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11511 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11512 Updated Vala bindings
11514 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11516 * gst/rtsp-server/rtsp-media.c:
11517 * gst/rtsp-server/rtsp-media.h:
11518 Added vmethod unprepare to GstRTSPMedia
11519 The default implementation sets the state of the pipeline to GST_STATE_NULL
11521 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11523 * gst/rtsp-server/rtsp-media-factory.c:
11524 * gst/rtsp-server/rtsp-media-factory.h:
11525 Made collect_streams function public
11527 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11529 * gst/rtsp-server/rtsp-media-factory.c:
11530 * gst/rtsp-server/rtsp-media-factory.h:
11531 * gst/rtsp-server/rtsp-media.c:
11532 Added vmethod create_pipeline to GstRTSPMediaFactory
11533 The pipeline is created in this method and the GstRTSPMedia's element is added to it
11535 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11537 * gst/rtsp-server/rtsp-client.c:
11538 client: use g_source_destroy()
11539 We need to use g_source_destroy() because we might have added the source to a
11540 different main context than the default one.
11542 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11544 * gst/rtsp-server/Makefile.am:
11545 * gst/rtsp-server/rtsp-client.c:
11546 * gst/rtsp-server/rtsp-params.c:
11547 * gst/rtsp-server/rtsp-params.h:
11548 rtsp: prepare for handling GET/SET_PARAMETER
11549 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
11551 Fix return codes of handlers.
11553 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11555 * gst/rtsp-server/rtsp-media.c:
11556 media: don't leak session pads
11558 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11560 * gst/rtsp-server/rtsp-media.c:
11561 media: clean up the messages a bit
11563 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11565 * gst/rtsp-server/rtsp-sdp.c:
11566 sdp: warn and skip streams without media
11568 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11570 * bindings/vala/gst-rtsp-server-0.10.vapi:
11571 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11572 vala: Fixed typo in header file of RTSPMediaStream
11574 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11576 * gst/rtsp-server/rtsp-media.c:
11578 Fix a debug message
11579 Make dumping RTCP stats configurable
11581 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11583 * gst/rtsp-server/rtsp-media.c:
11584 media: be less verbose and leak less
11586 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11588 * gst/rtsp-server/rtsp-media.c:
11589 media: don't leak the destination address
11591 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11593 * gst/rtsp-server/rtsp-client.c:
11594 * gst/rtsp-server/rtsp-media.c:
11595 * gst/rtsp-server/rtsp-media.h:
11596 * gst/rtsp-server/rtsp-session.c:
11597 * gst/rtsp-server/rtsp-session.h:
11598 rtsp: use RTCP to keep the session alive
11599 Use the RTCP rtcp-from stats field to find the associated session and use this
11600 to keep the session alive.
11602 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11604 * gst/rtsp-server/rtsp-session.c:
11605 session: add 5sec to the real session timeout
11606 Allow the session to live 5sec longer before really timing out. This should give
11607 clients some extra time to keep the session active.
11609 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11611 * gst/rtsp-server/rtsp-client.c:
11612 client: replay OK to GET/SET_PARAMETER
11613 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
11614 so that we return OK for those requests.
11616 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11618 * gst/rtsp-server/rtsp-media.c:
11619 * gst/rtsp-server/rtsp-media.h:
11620 media: keep track of active transports
11621 Keep track of which transport is active to avoid closing the connection too
11623 Remove the destination transport also when going to NULL.
11624 Print some stats about the SDES and other RTCP messages we receive from the
11627 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11629 * examples/.gitignore:
11630 * examples/Makefile.am:
11631 * examples/test-sdp.c:
11632 example: add SDP relay example
11634 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11636 * gst/rtsp-server/rtsp-media.c:
11637 media: also count active TCP connections
11639 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11641 * gst/rtsp-server/rtsp-media-factory.c:
11642 * gst/rtsp-server/rtsp-media.c:
11643 * gst/rtsp-server/rtsp-media.h:
11644 rtsp: add support for dynamic elements
11645 Add support for dynamic elements.
11646 Don't set live pipelines back to paused.
11648 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11650 * gst/rtsp-server/rtsp-sdp.c:
11651 sdp: don't add encoding name when absent in caps
11653 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11655 * gst/rtsp-server/rtsp-client.c:
11656 client: warn when we can't do RTP-Info
11658 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11660 * gst/rtsp-server/rtsp-media-factory.c:
11661 factory: factor out the stream construction
11663 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11665 * gst/rtsp-server/rtsp-client.c:
11666 client: only add RTP-Info when we have the info
11667 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
11670 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11673 back to development
11675 === release 0.10.3 ===
11677 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11681 - Fixes a bug where it put the wrong verion in pkgconfig
11682 - Link RTP and RTCP sources
11684 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11686 * gst/rtsp-server/rtsp-media.c:
11687 * gst/rtsp-server/rtsp-media.h:
11688 media: link the RTP udpsrc to the session manager
11689 Link the RTP udpsrc and the appsrc to the session manager so that they don't
11690 shut down when the client sends a packet to open firewalls.
11692 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11694 * pkgconfig/gst-rtsp-server.pc.in:
11695 Don't use hard-coded version number in pkg-config file
11697 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11700 back to development
11702 === release 0.10.2 ===
11704 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11709 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11712 * common/m4/.gitignore:
11713 * examples/.gitignore:
11714 * pkgconfig/.gitignore:
11715 add some .gitignore files
11717 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11719 * gst/rtsp-server/rtsp-media.c:
11720 media: seek to key frames
11722 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11724 * gst/rtsp-server/rtsp-media.c:
11725 media: emit the unprepared signal by id
11726 Emit the unprepared signal by id instead of name and set the media as
11729 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11731 * gst/rtsp-server/rtsp-media.c:
11732 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
11734 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11736 * gst/rtsp-server/rtsp-server.c:
11737 Added finalize function to GstRTPSPServer to unref session pool and media mapping
11739 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11741 * bindings/vala/gst-rtsp-server-0.10.vapi:
11742 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11743 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11744 Updated vala bindings
11746 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11748 * gst/rtsp-server/Makefile.am:
11749 * gst/rtsp-server/rtsp-client.c:
11750 * gst/rtsp-server/rtsp-media.c:
11751 server: use appsink and appsrc with the API
11752 Use the appsink/appsrc API instead of the signals for higher
11755 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11757 * examples/test-ogg.c:
11758 tests: set the payload type correctly
11760 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11762 * gst/rtsp-server/rtsp-media-factory.c:
11763 factory: connect to the unprepare signal
11764 Connect to the unprepare signal for non-reusable media so that we can remove
11765 them from the cache.
11767 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11769 * gst/rtsp-server/rtsp-media.c:
11770 * gst/rtsp-server/rtsp-media.h:
11771 media: add signal to notify of unprepare
11773 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11775 * gst/rtsp-server/rtsp-media.c:
11776 * gst/rtsp-server/rtsp-media.h:
11777 media: more work on making the media shared
11778 Add a reusable flag to medias, indicating that they can be reused after a state
11782 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11784 * examples/test-readme.c:
11785 examples: mark the example as shared for testing
11787 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11789 * gst/rtsp-server/rtsp-media.c:
11790 * gst/rtsp-server/rtsp-media.h:
11791 client: support shared media
11792 Always perform the state actions even if the target state of the pipeline is
11793 already correct, we still want to add/remove the transports when we are dealing
11795 Keep a counter of the number of active transports for a media so that we can use
11796 this to perform a state change when needed.
11797 Perform a state change of the pipeline only when the first transport was added
11798 or when there are no active transports.
11800 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11802 * gst/rtsp-server/rtsp-client.c:
11803 client: fix refcounting crasher
11804 Don't need to remove the weak refs in the finalize methods, they are already
11805 removed in the dispose.
11806 Don't register the callback with a DestroyNofity.
11808 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11810 * gst/rtsp-server/rtsp-client.c:
11811 Fix rtsp client refcount management in TCP mode.
11812 Don't unref a client ref we never had. Fixes an unref
11813 of an already-free client object after a client
11814 teardown request for me.
11816 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11818 * gst/rtsp-server/rtsp-session.c:
11819 docs: fix typo in API docs
11821 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11823 * gst/rtsp-server/rtsp-media.c:
11824 More seeking fixes.
11825 Keep the udp sources in playing even if we go to paused. unlock the sources when
11827 Add some more debug info.
11828 Only seek when we need to.
11829 Keep track of the position when we go to paused.
11831 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11833 * gst/rtsp-server/rtsp-client.c:
11834 * gst/rtsp-server/rtsp-media.c:
11835 * gst/rtsp-server/rtsp-media.h:
11836 Add beginnings of seeking.
11837 Parse the Range header and perform a seek on the pipeline for the requested
11838 position. It's disabled currently until I figure out what's going wrong.
11840 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11842 * gst/rtsp-server/rtsp-client.c:
11843 allow pause requests for now.
11846 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11848 * gst/rtsp-server/rtsp-client.c:
11849 Remove weak ref on the session in teardown
11850 We need to remove our weakref from the session when we do a teardown because
11851 else we close the TCP connection prematurely.
11853 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11855 * gst/rtsp-server/rtsp-client.c:
11856 * gst/rtsp-server/rtsp-client.h:
11857 * gst/rtsp-server/rtsp-session-pool.c:
11858 Do some more session cleanup
11859 Make session timeout kill the TCP connection that currently watches the
11861 Remove the client timeout property.
11863 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11865 * gst/rtsp-server/rtsp-client.c:
11866 * gst/rtsp-server/rtsp-client.h:
11867 * gst/rtsp-server/rtsp-media.c:
11868 * gst/rtsp-server/rtsp-media.h:
11869 * gst/rtsp-server/rtsp-server.c:
11870 * gst/rtsp-server/rtsp-session.c:
11871 * gst/rtsp-server/rtsp-session.h:
11873 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
11876 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11878 * examples/Makefile.am:
11879 * examples/test-launch.c:
11880 Add example server that takes launch lines
11881 Add an example server that streams any -launch line.
11883 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11885 * examples/test-readme.c:
11886 * gst/rtsp-server/rtsp-client.c:
11887 * gst/rtsp-server/rtsp-media.c:
11888 * gst/rtsp-server/rtsp-media.h:
11889 Add support for live streams
11890 Add support for live streams and ranges
11891 Start on handling TCP data transfer.
11893 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11895 * gst/rtsp-server/rtsp-media.c:
11896 Free the pipeline before other things
11899 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11901 * gst/rtsp-server/rtsp-client.c:
11902 Only free the pending tunnel if there is one
11905 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11907 * gst/rtsp-server/rtsp-client.c:
11908 * gst/rtsp-server/rtsp-client.h:
11909 * gst/rtsp-server/rtsp-media.c:
11910 rtsp-server: Add support for tunneling
11911 Add support for tunneling over HTTP.
11912 Use new connection methods to retrieve the url.
11913 Dispatch messages based on the message type instead of blindly
11914 assuming it's always a request.
11915 Keep track of the watch id so that we can remove it later.
11916 Set the media pipeline to NULL before unreffing the pipeline.
11918 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11920 * gst/rtsp-server/rtsp-client.c:
11921 * gst/rtsp-server/rtsp-client.h:
11922 Fix for channel -> watch rename in gstreamer
11923 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
11925 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11927 * gst/rtsp-server/rtsp-client.c:
11928 * gst/rtsp-server/rtsp-client.h:
11930 Use the async RTSP channels instead of spawning a new thread for each client.
11931 If a sessionid is specified in a request, fail if we don't have the session.
11933 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11935 * gst/rtsp-server/rtsp-media.c:
11936 Add better debug info
11937 Add some better debug info.
11939 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11941 * examples/test-video.c:
11943 Add support for session timeouts in the example.
11945 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11947 * gst/rtsp-server/rtsp-session-pool.c:
11948 * gst/rtsp-server/rtsp-session-pool.h:
11949 Pass GTimeVal around for performance reasons
11950 Get the current time only once and pass it around so that sessions don't have to
11951 get the current time anymore.
11952 Add experimental support for a GSource that dispatches when the session needs to
11955 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11957 * gst/rtsp-server/rtsp-session.c:
11958 * gst/rtsp-server/rtsp-session.h:
11959 Add better support for session timeouts
11960 Add a method to request the number of milliseconds when a session will timeout.
11962 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11964 * gst/rtsp-server/rtsp-media.c:
11965 * gst/rtsp-server/rtsp-media.h:
11966 Add suport for RTP manager monitoring
11967 Add the first stage in monitoring the rtp manager.
11968 Make sure we don't update the state to something we don't want.
11970 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11972 * gst/rtsp-server/rtsp-client.c:
11973 Add support for session keepalive
11974 Get and update the session timeout for all requests. get the session as early as
11977 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11979 * gst/rtsp-server/rtsp-media-factory.h:
11980 * gst/rtsp-server/rtsp-media.c:
11981 * gst/rtsp-server/rtsp-media.h:
11982 Handle media bus messages
11983 Handle media bus messages in a custom mainloop and dispatch them to the
11984 RTSPMedia objects. Let the default implementation handle some common messages.
11986 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11988 * gst/rtsp-server/rtsp-client.c:
11989 * gst/rtsp-server/rtsp-session-pool.c:
11990 * gst/rtsp-server/rtsp-session.c:
11991 Some more session timeout handling
11992 Move the session header setting code to a central place so that we always add
11993 the timeout parameter too.
11994 Handle timeouts by running the session cleanup code.
11995 Stop media before cleaning up.
11997 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11999 * gst/rtsp-server/rtsp-client.c:
12000 * gst/rtsp-server/rtsp-client.h:
12001 Add timeout property
12002 Add a timeout property ot the client and make the other properties into GObject
12005 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12007 * gst/rtsp-server/rtsp-session-pool.c:
12008 Use getters and setters in property code
12009 Use the getters and setters for the timeout property instead of locking
12012 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12014 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
12016 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12018 * gst/rtsp-server/rtsp-session-pool.c:
12019 * gst/rtsp-server/rtsp-session-pool.h:
12020 * gst/rtsp-server/rtsp-session.c:
12021 * gst/rtsp-server/rtsp-session.h:
12022 Add more timeout stuff
12023 Add method to check if a session is expired.
12024 Add method to perform cleanup on a session pool.
12026 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12028 * gst/rtsp-server/rtsp-client.c:
12029 * gst/rtsp-server/rtsp-session-pool.c:
12030 * gst/rtsp-server/rtsp-session-pool.h:
12031 * gst/rtsp-server/rtsp-session.c:
12032 * gst/rtsp-server/rtsp-session.h:
12033 Add beginnings of session timeouts and limits
12034 Add the timeout value to the Session header for unusual timeout values.
12035 Allow us to configure a limit to the amount of active sessions in a pool. Set a
12036 limit on the amount of retry we do after a sessionid collision.
12037 Add properties to the sessionid and the timeout of a session. Keep track of
12038 creation time and last access time for sessions.
12040 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12042 * gst/rtsp-server/rtsp-client.c:
12043 * gst/rtsp-server/rtsp-media.c:
12044 * gst/rtsp-server/rtsp-media.h:
12045 * gst/rtsp-server/rtsp-sdp.c:
12046 * gst/rtsp-server/rtsp-session-pool.c:
12047 * gst/rtsp-server/rtsp-session.c:
12048 * gst/rtsp-server/rtsp-session.h:
12049 Cleanup of sessions and more
12050 Fix the refcounting of media and sessions in the client. Properly clean up the
12051 session data when the client performs a teardown.
12052 Add Server header to responses.
12053 Allow for multiple uri setups in one session.
12054 Add Range header to the PLAY response and add the range attribute to the SDP
12056 Fix the session pool remove method, it used the wrong key in the hashtable. Also
12057 give the ownership of the sessionid to the session object.
12059 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12061 * gst/rtsp-server/rtsp-server.c:
12062 * gst/rtsp-server/rtsp-server.h:
12064 Rename the 'server_port' variable to simply 'port'.
12066 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12069 * gst/rtsp-server/rtsp-client.c:
12070 * gst/rtsp-server/rtsp-media.c:
12071 * gst/rtsp-server/rtsp-media.h:
12072 * gst/rtsp-server/rtsp-session.c:
12073 * gst/rtsp-server/rtsp-session.h:
12074 Rework the way we handle transports for streams
12075 Make the media accept an array of transports for the streams that we have
12076 configured for the play/pause requests.
12077 Implement server states for a client and its media.
12078 Require 0.10.22.1 (git HEAD) of gstreamer.
12080 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12082 * gst/rtsp-server/rtsp-client.c:
12083 * gst/rtsp-server/rtsp-media-factory.c:
12084 Drop const from functions dealing with urls
12085 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
12086 have the right const in them.
12088 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12090 * gst/rtsp-server/rtsp-client.c:
12091 * gst/rtsp-server/rtsp-media.c:
12092 * gst/rtsp-server/rtsp-sdp.c:
12096 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12098 * gst/rtsp-server/rtsp-client.c:
12099 * gst/rtsp-server/rtsp-media-factory.c:
12100 * gst/rtsp-server/rtsp-media.c:
12101 * gst/rtsp-server/rtsp-media.h:
12103 Don't keep a reference to the GstRTSPMedia in the stream.
12104 Free more things when freeing the GstRTSPMedia.
12106 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12109 * gst/rtsp-server/rtsp-media-factory.c:
12110 * gst/rtsp-server/rtsp-media-factory.h:
12111 * gst/rtsp-server/rtsp-media.c:
12112 * gst/rtsp-server/rtsp-media.h:
12113 * gst/rtsp-server/rtsp-server.c:
12114 * gst/rtsp-server/rtsp-server.h:
12115 More docs and small cleanups
12116 Add some more docs and update the README
12117 Cleanup some method names.
12118 Remove an unneeded idx field in the GstRTSPMediaStream
12120 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12123 * examples/Makefile.am:
12124 * examples/test-readme.c:
12125 Add a README and more example code
12126 Add a README file that contains a small introduction on how to use the server
12127 along with the example code explained in the readme.
12129 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12131 * gst/rtsp-server/rtsp-media.c:
12132 * gst/rtsp-server/rtsp-server.c:
12133 Fix some leaks and change default port
12134 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
12135 we finished the initial preroll. If we keep them locked, setting the pipeline to
12136 NULL will not stop and clean up the sources correctly.
12137 Change the default RTSP port to 8554 aka the official alternative RTSP port.
12139 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12141 * gst/rtsp-server/rtsp-session.c:
12142 * gst/rtsp-server/rtsp-session.h:
12143 Cleanups to the session object
12144 Remove some unneeded variables in the session state of a stream such as the
12145 owner media and the server transport.
12146 Get the configuration of a media stream in a session based on the media_stream
12147 in the original object instead of our cached index.
12148 Free more data in the finalize method.
12150 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12152 * gst/rtsp-server/rtsp-client.c:
12153 * gst/rtsp-server/rtsp-client.h:
12154 Cleanups and reuse media from DESCRIBE
12155 Handle thread create errors.
12156 Rename some internal methods to better match what they actually do.
12157 Handle misconfiguration of session_pool and media_mapping gracefully.
12158 Cache the DESCRIBE media and uri in the client connection and reuse them when
12159 we receive a SETUP request in the same connection for the same uri.
12160 Cleanup the client connection object.
12162 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12164 * gst/rtsp-server/rtsp-media-factory.c:
12165 * gst/rtsp-server/rtsp-media-factory.h:
12166 * gst/rtsp-server/rtsp-media.c:
12167 * gst/rtsp-server/rtsp-media.h:
12168 Add shared properties to media and factory
12169 Add the shared property to media.
12170 Implement some simple caching in the factory depending on if the media is shared
12173 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12175 * gst/rtsp-server/rtsp-client.c:
12176 Add a little comment
12177 Add some comment about the content-base header.
12179 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12181 * examples/Makefile.am:
12182 * examples/test-mp4.c:
12183 * examples/test-ogg.c:
12184 * examples/test-video.c:
12185 * gst/rtsp-server/Makefile.am:
12186 * gst/rtsp-server/rtsp-client.c:
12187 * gst/rtsp-server/rtsp-client.h:
12188 * gst/rtsp-server/rtsp-media-factory.c:
12189 * gst/rtsp-server/rtsp-media-factory.h:
12190 * gst/rtsp-server/rtsp-media.c:
12191 * gst/rtsp-server/rtsp-media.h:
12192 * gst/rtsp-server/rtsp-sdp.c:
12193 * gst/rtsp-server/rtsp-sdp.h:
12194 * gst/rtsp-server/rtsp-server.c:
12195 * gst/rtsp-server/rtsp-server.h:
12196 * gst/rtsp-server/rtsp-session.c:
12197 * gst/rtsp-server/rtsp-session.h:
12198 Reorganize things, prepare for media sharing
12199 Added various other test server examples
12200 Move the SDP message generation to a separate helper.
12201 Refactor common code for finding the session.
12202 Add content-base for realplayer compatibility
12203 Clean up request uris before processing for better vlc compatibility.
12204 Move prerolling and pipeline construction to the RTSPMedia object.
12205 Use multiudpsink for future pipeline reuse.
12207 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12210 Back to development
12213 === release 0.10.1 ===
12215 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12218 Make 0.10.1 release
12221 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12223 * bindings/vala/Makefile.am:
12225 Add more directories and files to the dist.
12227 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12229 * bindings/python/Makefile.am:
12230 * bindings/python/rtspserver.override:
12231 Fixed compile error of python bindings
12233 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12235 * bindings/vala/gst-rtsp-server-0.10.vapi:
12236 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12237 Marked values as nullable accordingly
12239 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12241 * bindings/vala/gst-rtsp-server-0.10.vapi:
12242 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12243 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12244 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12245 Updated Vala bindings
12247 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12249 * gst/rtsp-server/rtsp-client.c:
12250 * gst/rtsp-server/rtsp-media-mapping.c:
12251 * gst/rtsp-server/rtsp-media-mapping.h:
12252 * gst/rtsp-server/rtsp-media.h:
12253 * gst/rtsp-server/rtsp-session-pool.h:
12254 Cleanups and doc updates
12255 Add some more documentation and do some minor cleanups here and there.
12257 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12259 * gst/rtsp-server/rtsp-client.c:
12260 * gst/rtsp-server/rtsp-media-factory.c:
12261 * gst/rtsp-server/rtsp-media-factory.h:
12262 * gst/rtsp-server/rtsp-media.c:
12263 * gst/rtsp-server/rtsp-media.h:
12264 * gst/rtsp-server/rtsp-session.c:
12265 * gst/rtsp-server/rtsp-session.h:
12267 Rename GstRTSPMediaBin to GstRTSPMedia
12268 Parse the request url into a GstRTSPUri object and pass this object to the
12269 various handlers and methods that require the uri.
12271 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12275 Add some more docs and remove some old code from the example.
12277 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12279 * gst/rtsp-server/rtsp-client.c:
12280 Handle state change failures better
12281 Handle state change failures better when changing the state of the pipeline to
12284 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12286 * gst/rtsp-server/rtsp-media-factory.c:
12287 * gst/rtsp-server/rtsp-media-factory.h:
12288 Make element creation more extendible
12289 Add get_element vmethod to the default MediaFactory so that subclasses can just
12290 override that method and still use the default logic for making a MediaBin from
12293 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12296 * gst/rtsp-server/Makefile.am:
12297 * gst/rtsp-server/rtsp-client.c:
12298 * gst/rtsp-server/rtsp-client.h:
12299 * gst/rtsp-server/rtsp-media-factory.c:
12300 * gst/rtsp-server/rtsp-media-factory.h:
12301 * gst/rtsp-server/rtsp-media-mapping.c:
12302 * gst/rtsp-server/rtsp-media-mapping.h:
12303 * gst/rtsp-server/rtsp-media.c:
12304 * gst/rtsp-server/rtsp-media.h:
12305 * gst/rtsp-server/rtsp-server.c:
12306 * gst/rtsp-server/rtsp-server.h:
12307 * gst/rtsp-server/rtsp-session.c:
12308 * gst/rtsp-server/rtsp-session.h:
12309 Make the server handle arbitrary pipelines
12310 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
12311 The GstMediaBin object has a handle to a bin with elements and to a list of
12312 GstMediaStream objects that this bin produces.
12313 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
12314 with methods to register and remove those mappings.
12315 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
12316 used by the server instance.
12317 Modify the example application so that it shows how to create custom pipelines
12318 attached to a specific mount point.
12319 Various misc cleanps.
12321 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12323 * gst/rtsp-server/rtsp-server.c:
12324 * gst/rtsp-server/rtsp-server.h:
12325 Allow setting a custom media factory for a server
12327 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12329 * gst/rtsp-server/rtsp-client.c:
12330 * gst/rtsp-server/rtsp-client.h:
12331 Allow setting a custom media factory for a client.
12333 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12335 * gst/rtsp-server/Makefile.am:
12336 Add Makefile entry for the media factory
12338 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12340 * gst/rtsp-server/rtsp-media-factory.c:
12341 * gst/rtsp-server/rtsp-media-factory.h:
12342 Add media factory to map urls to media pipeline objects.
12344 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12346 * gst/rtsp-server/rtsp-media.c:
12347 * gst/rtsp-server/rtsp-media.h:
12348 Add comments. Remove unused field
12350 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12352 * gst/rtsp-server/rtsp-session-pool.c:
12353 * gst/rtsp-server/rtsp-session-pool.h:
12354 Allow custom session pools to override the session id allocation algorithms Add some comments.
12356 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12358 * gst/rtsp-server/rtsp-session.h:
12361 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12363 * gst/rtsp-server/rtsp-client.c:
12364 * gst/rtsp-server/rtsp-client.h:
12365 Move the connection code in one place Add some comments
12367 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12369 * gst/rtsp-server/rtsp-server.c:
12370 * gst/rtsp-server/rtsp-server.h:
12371 Make vmethod to create and accept new clients. Add some docs.
12373 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12375 * gst/rtsp-server/rtsp-server.c:
12376 * gst/rtsp-server/rtsp-server.h:
12377 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
12379 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12381 * gst/rtsp-server/rtsp-client.c:
12382 * gst/rtsp-server/rtsp-client.h:
12383 Name the parameters more appropriately.
12385 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12387 * gst/rtsp-server/rtsp-session-pool.c:
12388 Do some more cleanup of the session pool.
12390 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12392 * gst/rtsp-server/Makefile.am:
12393 * gst/rtsp-server/rtsp-client.c:
12394 Check if return value of gst_rtsp_session_get_media is not NULL
12396 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12398 * gst/rtsp-server/Makefile.am:
12399 Install rtsp-session and rtsp-session-pool headers
12401 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12406 * bindings/python/Makefile.am:
12407 * bindings/python/arg-types.py:
12408 * bindings/python/codegen/Makefile.am:
12409 * bindings/python/codegen/__init__.py:
12410 * bindings/python/codegen/argtypes.py:
12411 * bindings/python/codegen/code-coverage.py:
12412 * bindings/python/codegen/codegen.py:
12413 * bindings/python/codegen/definitions.py:
12414 * bindings/python/codegen/defsparser.py:
12415 * bindings/python/codegen/docextract.py:
12416 * bindings/python/codegen/docgen.py:
12417 * bindings/python/codegen/fileprefix.override:
12418 * bindings/python/codegen/fileprefixmodule.c:
12419 * bindings/python/codegen/h2def.py:
12420 * bindings/python/codegen/mergedefs.py:
12421 * bindings/python/codegen/mkskel.py:
12422 * bindings/python/codegen/override.py:
12423 * bindings/python/codegen/reversewrapper.py:
12424 * bindings/python/codegen/scmexpr.py:
12425 * bindings/python/rtspserver-types.defs:
12426 * bindings/python/rtspserver.defs:
12427 * bindings/python/rtspserver.override:
12428 * bindings/python/rtspservermodule.c:
12430 Add python bindings.
12432 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12434 * bindings/Makefile.am:
12436 Don't go into python dir when requirements for python bindings are missing
12438 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12440 * bindings/Makefile.am:
12441 * bindings/vala/Makefile.am:
12443 Install Vala bindings if vala is available
12445 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12447 * bindings/vala/gst-rtsp-server-0.10.deps:
12448 * bindings/vala/gst-rtsp-server-0.10.vapi:
12449 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
12450 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12451 * bindings/vala/packages/gst-rtsp-server-0.10.files:
12452 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12453 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12454 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
12455 Regenerated Vala bindings
12457 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12459 * bindings/vala/gst-rtsp-server.vapi:
12460 * bindings/vala/packages/gst-rtsp-server.metadata:
12461 Fixed typo in included headers for vala bindings
12463 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12467 * pkgconfig/Makefile.am:
12468 * pkgconfig/gst-rtsp-server.pc.in:
12469 Added pkgconfig file
12471 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12473 * bindings/vala/gst-rtsp-server.vapi:
12474 * bindings/vala/packages/gst-rtsp-server.excludes:
12475 * bindings/vala/packages/gst-rtsp-server.gi:
12476 * bindings/vala/packages/gst-rtsp-server.metadata:
12477 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
12479 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12481 * bindings/vala/gst-rtsp-server.vapi:
12482 * bindings/vala/packages/gst-rtsp-server.deps:
12483 * bindings/vala/packages/gst-rtsp-server.files:
12484 * bindings/vala/packages/gst-rtsp-server.gi:
12485 * bindings/vala/packages/gst-rtsp-server.metadata:
12486 * bindings/vala/packages/gst-rtsp-server.namespace:
12487 Added Vala bindings
12489 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
12491 * gst/rtsp-server/rtsp-session.c:
12492 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
12494 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12496 * examples/Makefile.am:
12497 * gst/rtsp-server/Makefile.am:
12498 Put GStreamer version in library name
12500 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12502 * examples/Makefile.am:
12503 * gst/rtsp-server/Makefile.am:
12504 Fix some issues to pass distcheck
12506 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12508 * gst/rtsp-server/rtsp-server.c:
12509 Added port property to GstRTSPServer class.
12511 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12516 * examples/Makefile.am:
12519 * gst/rtsp-server/Makefile.am:
12520 * gst/rtsp-server/rtsp-client.c:
12521 * gst/rtsp-server/rtsp-client.h:
12522 * gst/rtsp-server/rtsp-media.c:
12523 * gst/rtsp-server/rtsp-media.h:
12524 * gst/rtsp-server/rtsp-server.c:
12525 * gst/rtsp-server/rtsp-server.h:
12526 * gst/rtsp-server/rtsp-session-pool.c:
12527 * gst/rtsp-server/rtsp-session-pool.h:
12528 * gst/rtsp-server/rtsp-session.c:
12529 * gst/rtsp-server/rtsp-session.h:
12531 Split in library and example program
12533 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12535 * src/rtsp-client.h:
12536 Removed obsolete variable
12538 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12540 * src/rtsp-client.c:
12541 * src/rtsp-client.h:
12542 Removed pipeline variable GstRTSPClient, because it's only used in one function
12544 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12546 * src/rtsp-media.c:
12547 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
12549 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
12551 * src/rtsp-session.c:
12552 Initialize some more vars.
12554 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
12556 * src/rtsp-session.c:
12557 Initialize variable to avoid compiler warning.
12559 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
12562 Add a reasonable generic .gitignore