3 2015-08-19 Sebastian Dröge <slomo@coaxion.net>
8 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
10 * gst/rtsp-server/rtsp-media-factory.c:
11 media-factory: get port number through gst_rtsp_url_get_port
12 https://bugzilla.gnome.org/show_bug.cgi?id=753473
14 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
16 * tests/check/gst/media.c:
17 media-test: Removing unnecessary assertion
18 https://bugzilla.gnome.org/show_bug.cgi?id=753385
20 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
22 * gst/rtsp-server/rtsp-server.c:
23 Document that source keeps a ref on server until it's destroyed
24 https://bugzilla.gnome.org/show_bug.cgi?id=749227
26 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
28 * tests/check/gst/media.c:
29 media-test: Test for multiple dynamic payload
30 https://bugzilla.gnome.org/show_bug.cgi?id=753385
32 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
34 * gst/rtsp-server/rtsp-media.c:
35 media: Only add fakesink once per pipeline
36 The intention is to prevent going PLAYING state before pads are created.
37 If there was mutilple dynamic payload, it would leak few fakesink and
38 actually prevent from ever reaching playing state.
39 https://bugzilla.gnome.org/show_bug.cgi?id=753385
41 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
43 * gst/rtsp-server/rtsp-media.c:
44 Revert "rtsp-media: Only add 1 fakesink per pipeline"
45 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
47 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
49 * gst/rtsp-server/rtsp-media.c:
50 rtsp-media: Only add 1 fakesink per pipeline
51 There should be only one fakesink per pipeline, not per dynpay. This
52 would lead to element naming clash.
54 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
56 * gst/rtsp-server/rtsp-media.c:
57 rtsp-media: assertion error due to wrong condition check
58 In media to caps function, reserved_keys array is being used for variable i,
59 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
60 changed it to variable j
61 https://bugzilla.gnome.org/show_bug.cgi?id=753009
63 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
65 * gst/rtsp-server/rtsp-media.c:
66 rtsp-media: Strip keys from the fmtp that we use internally in our caps
67 Skip keys from the fmtp, which we already use ourselves for the
68 caps. Some software is adding random things like clock-rate into
69 the fmtp, and we would otherwise here set a string-typed clock-rate
70 in the caps... and thus fail to create valid RTP caps
71 https://bugzilla.gnome.org/show_bug.cgi?id=753009
73 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
75 * gst/rtsp-server/rtsp-thread-pool.c:
76 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
77 https://bugzilla.gnome.org/show_bug.cgi?id=752640
79 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
82 Automatic update of common submodule
83 From f74b2df to 9aed1d7
85 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
92 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
98 * gst-rtsp-server.doap:
101 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
103 * gst/rtsp-server/rtsp-client.c:
104 * gst/rtsp-server/rtsp-client.h:
105 * tests/check/gst/client.c:
106 rtsp-client: allow application to decide what requirements are supported
107 Add "check-requirements" signal and vfunc to allow application
108 (and subclasses) to check the requirements.
109 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
110 https://bugzilla.gnome.org/show_bug.cgi?id=749417
112 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
115 Automatic update of common submodule
116 From 6015d26 to f74b2df
118 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
120 * gst/rtsp-server/rtsp-media.c:
121 rtsp-media: Always use real payloader when creating streams
122 A bin that contains the real payloader might be used as payloader. In this
123 case we have to get the real payloader for the various properties it provides.
124 Example use cases for this are bins that payload some media and then have
125 additional elements that add metadata or RTP extension headers to the stream.
126 https://bugzilla.gnome.org/show_bug.cgi?id=750800
128 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
130 * examples/test-netclock-client.c:
131 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
133 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
135 * examples/test-netclock-client.c:
136 * examples/test-netclock.c:
137 test-netclock: Use new ntp-time-source property on rtpbin
138 Select the clock time to be used as NTP time source. This allows proper
139 synchronization between receivers, independent of sharing base times, and just
140 requires them to use the same clock.
142 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
144 * examples/test-netclock-client.c:
145 * examples/test-netclock.c:
146 test-netclock: Setting the same base time on sender and receiver is not necessary
147 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
149 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
151 * gst/rtsp-server/rtsp-stream.c:
152 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
153 https://bugzilla.gnome.org/show_bug.cgi?id=750764
155 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
157 * docs/libs/gst-rtsp-server.types:
158 docs: add missing types
159 https://bugzilla.gnome.org/show_bug.cgi?id=750764
161 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
163 * docs/libs/gst-rtsp-server-sections.txt:
164 docs: add missing apis
165 https://bugzilla.gnome.org/show_bug.cgi?id=750764
167 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
169 * examples/test-netclock-client.c:
170 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
172 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
174 * docs/libs/gst-rtsp-server-sections.txt:
175 * gst/rtsp-server/rtsp-auth.c:
176 * gst/rtsp-server/rtsp-auth.h:
177 GstRTSPAuth: Add client certificate authentication support
178 https://bugzilla.gnome.org/show_bug.cgi?id=750471
180 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
182 * examples/test-netclock-client.c:
183 test-netclock-client: Use new GstClock API to wait for clock synchronization
185 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
187 * examples/test-netclock-client.c:
188 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
189 A mainloop is needed to get glimagesink to display something on OSX, and
190 the source-setup signal just makes things a little bit easier.
192 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
195 Automatic update of common submodule
196 From d9a3353 to 6015d26
198 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
201 Automatic update of common submodule
202 From d37af32 to d9a3353
204 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
207 Automatic update of common submodule
208 From 21ba2e5 to d37af32
210 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
213 Automatic update of common submodule
214 From c408583 to 21ba2e5
216 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
218 * docs/libs/Makefile.am:
219 docs: remove variables that we define in the snippet from common
220 This is syncing our Makefile.am with upstream gtkdoc.
222 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
225 Automatic update of common submodule
226 From 44a3517 to c408583
228 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
233 === release 1.5.1 ===
235 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
241 * gst-rtsp-server.doap:
244 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
246 * gst/rtsp-server/rtsp-client.c:
247 rtsp-client: No flush during Teardown.
248 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
249 backlog is empty it can happen that just a part of a message will be
250 sent and rest is in backlog queue. If then flush during teardown
251 just a part of message will be sent.This can lead to client miss
252 teardown response since it expect to get the last part of message.
253 The flushing during teardown was introduced to fix a deadlock that now
254 is fixed more generally in handle_request by temporary setting backlog
256 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
258 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
260 * tests/check/Makefile.am:
261 tests: Use AM_TESTS_ENVIRONMENT
262 Needed by the new automake test runner and the
263 current version of the common submodule.
265 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
267 * gst/rtsp-server/rtsp-media.h:
268 * gst/rtsp-server/rtsp-stream.h:
269 rtsp-server: Use single-include rtsp header to make sure we get all definitions
271 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
273 * gst/rtsp-server/rtsp-media.c:
274 rtsp-media: Mark some more functions static
276 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
278 * gst/rtsp-server/rtsp-media.c:
279 rtsp-media: Only unblock the media in suspend() when actually changing the state
280 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
282 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
284 * examples/test-video-rtx.c:
285 examples: Use AVPF profile for the RTX example
287 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
289 * gst/rtsp-server/rtsp-sdp.c:
290 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
292 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
294 * gst/rtsp-server/rtsp-stream.c:
295 rtsp-stream: get valid clock-rate from last-sample
296 clock-rate in last-sample's caps is integer, not unsigned.
297 To get this value properly, variable needs to be type-casted to int.
298 https://bugzilla.gnome.org/show_bug.cgi?id=747614
300 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
304 autogen.sh: only run autopoint if gettext requested in configure.ac
305 Not just because there happens to be a po directory.
306 https://bugzilla.gnome.org/show_bug.cgi?id=748058
308 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
311 Revert "configure.ac: uncomment gettext version setup"
312 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
313 We don't need a gettext setup here and there's no po
314 directory either, so no reason why autopoint would be
315 run in the first place.
316 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
318 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
320 * examples/test-multicast.c:
321 * examples/test-multicast2.c:
322 * examples/test-sdp.c:
323 * examples/test-video-rtx.c:
324 * examples/test-video.c:
325 * tests/test-cleanup.c:
326 * tests/test-reuse.c:
327 Fix timeout function signatures across tests and examples
329 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
331 * tests/check/Makefile.am:
332 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
333 Make sure the test environment is set up.
334 https://bugzilla.gnome.org//show_bug.cgi?id=747624
336 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
339 configure: bump automake requirement to 1.14 and autoconf to 2.69
340 This is only required for builds from git, people can still
341 build tarballs if they only have older autotools.
342 https://bugzilla.gnome.org//show_bug.cgi?id=747624
344 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
347 configure.ac: uncomment gettext version setup
348 Fixes autogen.sh. It would run autopoint, which would complain
349 that it could not find the gettext version in configure.ac.
350 https://bugzilla.gnome.org/show_bug.cgi?id=748058
352 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
354 * examples/test-video-rtx.c:
355 test-video-rtx: set exact payload type to PCMA payloader
356 Setting wrong payload type causes failure to do retransmission through audio stream
357 https://bugzilla.gnome.org/show_bug.cgi?id=747839
359 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
361 * gst/rtsp-server/rtsp-media.c:
362 * gst/rtsp-server/rtsp-stream.c:
363 * gst/rtsp-server/rtsp-stream.h:
364 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
365 Because of duplicated g_signal_connect for request-aux-sender signal,
366 wrong stream pointer is passed to the signal handler.
367 Instead of passing each stream, pass stream array and get the relevant stream.
368 https://bugzilla.gnome.org/show_bug.cgi?id=747839
370 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
374 Update autogen.sh to latest version from common
375 Fixes build after aclocal_check etc. helpers have been removed.
377 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
380 Automatic update of common submodule
381 From bc76a8b to c8fb372
383 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
385 * gst/rtsp-server/rtsp-stream.c:
386 rtsp-stream: Limit the queues to 1 buffer
387 We only need them to be able to pre-roll, queueing up more data here
388 is only going to harm latency and memory usage.
390 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
392 * gst/rtsp-server/rtsp-stream.c:
393 rtsp-stream: Update comment and ASCII art to the latest code
394 We have a queue in front of the udpsink too to prevent the pipeline from
397 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
399 * gst/rtsp-server/rtsp-stream.c:
400 rtsp-media: Properly return first rtptime
401 Instead we where returning first GstBuffer timestamp. This would result
402 in clock skew and unwanted behaviour in RTSP playback.
403 https://bugzilla.gnome.org/show_bug.cgi?id=746479
405 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
407 * gst/rtsp-server/rtsp-stream.c:
408 rtsp-stream: Don't leave buffer mapped
409 If the seq is NULL, the RTP buffer was left mapped. We should always
412 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
417 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
419 * gst/rtsp-server/rtsp-media-factory.c:
420 * tests/check/gst/client.c:
421 Fix double semicolons
423 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
425 * gst/rtsp-server/rtsp-stream.c:
426 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
427 This gives more accurate values than asking the payloader. There might be
428 queueing happening between the payloader and the sink.
429 https://bugzilla.gnome.org/show_bug.cgi?id=745704
431 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
433 * gst/rtsp-server/rtsp-media.c:
434 rtsp-media: Don't seek for PLAY if the position will not change
435 https://bugzilla.gnome.org/show_bug.cgi?id=745704
437 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
439 * gst/rtsp-server/rtsp-media.c:
440 rtsp-media: Don't include payload type in the caps for framesize
441 When the sdp media attribute framesize are converted to caps
442 the <payload> should not be included.
443 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
444 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
446 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
448 * gst/rtsp-server/rtsp-sdp.c:
449 rtsp-sdp: add payload type to the sdp framesize attribute
450 The sdp framesize attribute is desribed in RFC6064. It is specified
451 for payloading of H263 and has the following form
452 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
453 should be added to the caps in a payloader and the <payload type> should
454 be added by the rtsp-server.
455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
457 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
459 * examples/test-uri.c:
460 examples: test-uri: fix tainted variable
461 Insignificant but this keeps Coverity happy.
464 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
466 * examples/.gitignore:
467 * examples/Makefile.am:
468 * examples/test-netclock-client.c:
469 * examples/test-netclock.c:
470 examples: Add a simple example of network synch for live streams.
471 An example server and client that works for synchronising live streams
472 only - as it can't support pause/play.
474 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
476 * gst/rtsp-server/rtsp-media-factory.c:
477 * gst/rtsp-server/rtsp-media-factory.h:
478 rtsp-media-factory: Add functions to set/get the media gtype
479 Allow specifying the GType of a GstRtspMedia subclass to create
480 as a simpler way to get the factory to create a custom
481 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
483 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
485 * gst/rtsp-server/rtsp-media.c:
486 rtsp-media: fix double unlock in _get_buffer_size()
487 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
488 because of double g_mutex_unlock () usage.
489 https://bugzilla.gnome.org/show_bug.cgi?id=745434
491 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
493 * gst/rtsp-server/rtsp-session-pool.c:
494 * gst/rtsp-server/rtsp-session.c:
495 * gst/rtsp-server/rtsp-session.h:
496 rtsp-session: Use monotonic time for RTSP session timeout
497 Changed RTSP session timeout handling to monotonic time
498 and deprecating the API for current system time.
499 This fixes timeouts when the system time changes.
500 https://bugzilla.gnome.org/show_bug.cgi?id=743346
502 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
504 * gst/rtsp-server/rtsp-client.c:
505 * gst/rtsp-server/rtsp-media.c:
506 rtsp-client: Only error out in PLAY if seeking actually failed
507 If the media was just not seekable, we continue from whatever position we are
508 and let the client decide if that is what is wanted or not.
509 Only if the actual seek failed, we can't really recover and should error out.
511 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
513 * gst/rtsp-server/rtsp-stream.c:
514 rtsp-stream: Add necessary queues between tee and multiudpsink
515 https://bugzilla.gnome.org/show_bug.cgi?id=744379
517 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
519 * gst/rtsp-server/rtsp-client.c:
520 * gst/rtsp-server/rtsp-media.c:
521 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
522 Instead error out properly the same way as if the SEEKING query already
525 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
527 * gst/rtsp-server/rtsp-stream.h:
528 rtsp-stream: minor code formatting fix
530 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
532 * gst/rtsp-server/rtsp-media.c:
533 rtsp-media: fix logic for collect_streams
534 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
535 all streams it knows if it got any, and can check if the transport mode is OK.
538 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
540 * gst/rtsp-server/rtsp-media.c:
541 rtsp-media: Don't set the transport mode based on what elements we find
542 Just print a warning if the one that was set before disagrees with what
543 elements we found. It must already be set to something before as this
544 function is called after we received the SDP from ANNOUNCE in RECORD mode,
545 and we would reject ANNOUNCE if the RECORD flag was not set.
547 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
549 * tests/check/gst/rtspserver.c:
550 tests: rtspserver: rename shadowed variable
551 We have two different 'sink' variables here,
552 rename one of them for clarity.
554 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
556 * gst/rtsp-server/rtsp-client.c:
557 rtsp-client: fix awkward if clause
559 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
561 * examples/test-uri.c:
562 examples: test-uri: improve uri argument handling and accept file names
563 Print an error if the argument passed is not a URI and can't
564 be converted into one, or no arguments have been provided.
566 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
568 * examples/test-uri.c:
569 examples: test-uri: don't remove mount point after 10 seconds
570 It's very irritating when trying to test stuff repeatedly
571 and serves no real purpose other than showing that it can
574 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
576 * examples/.gitignore:
577 examples: add new test-record to .gitignore
579 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
581 * examples/test-record.c:
582 * gst/rtsp-server/rtsp-client.c:
583 * gst/rtsp-server/rtsp-media-factory.c:
584 * gst/rtsp-server/rtsp-media-factory.h:
585 * gst/rtsp-server/rtsp-media.c:
586 * gst/rtsp-server/rtsp-media.h:
587 * tests/check/gst/rtspserver.c:
588 rtsp-media: Use flags to distinguish between PLAY and RECORD media
590 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
592 * examples/test-record.c:
593 test-record: Set latency for playback-style example to 2s instead of 200ms
595 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
597 * tests/check/gst/rtspserver.c:
598 tests: add some unit tests for ANNOUNCE and RECORD
599 https://bugzilla.gnome.org/show_bug.cgi?id=743175
601 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
603 * gst/rtsp-server/rtsp-client.c:
604 rtsp-client: fix a couple of leaks in handle_announce
606 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
608 * gst/rtsp-server/rtsp-media-factory.c:
609 * gst/rtsp-server/rtsp-media-factory.h:
610 * gst/rtsp-server/rtsp-media.c:
611 * gst/rtsp-server/rtsp-media.h:
612 rtsp-media: Expose latency setting for setting the rtpbin latency
614 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
616 * examples/test-record.c:
617 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
619 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
621 * gst/rtsp-server/rtsp-stream.c:
622 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
624 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
626 * examples/Makefile.am:
627 * examples/test-record.c:
628 * gst/rtsp-server/rtsp-client.c:
629 * gst/rtsp-server/rtsp-client.h:
630 * gst/rtsp-server/rtsp-media-factory.c:
631 * gst/rtsp-server/rtsp-media-factory.h:
632 * gst/rtsp-server/rtsp-media.c:
633 * gst/rtsp-server/rtsp-media.h:
634 * gst/rtsp-server/rtsp-session-media.c:
635 * gst/rtsp-server/rtsp-stream.c:
636 * gst/rtsp-server/rtsp-stream.h:
637 Add initial support for RECORD
638 We currently only support media that is RECORD or PLAY only, not both at once.
639 https://bugzilla.gnome.org/show_bug.cgi?id=743175
641 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
643 * gst/rtsp-server/rtsp-stream.c:
644 rtsp-stream: RTCP and RTP transport cache cookies seperated
645 RTCP packets were not sent because the same tr_cache_cookie was used for
646 both RTP and RTCP. So only one of the tr_cache lists were populated
647 depending on which one was sent first. If the tr_cache list is not
648 populated then no packets can be sent. Most often this happened to be
649 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
650 resulted in both the tr_cache_lists to be populated regardless of which
652 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
654 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
656 * gst/rtsp-server/rtsp-stream.c:
657 rtsp-stream: fix false compiler warning
658 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
660 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
662 * gst/rtsp-server/rtsp-client.c:
663 rtsp-client: log interleaved data received
665 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
667 * gst/rtsp-server/rtsp-client.c:
668 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
670 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
672 * gst/rtsp-server/rtsp-client.c:
673 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
675 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
677 * gst/rtsp-server/rtsp-client.c:
678 rtsp-client: Use a random session ID in the SDP
679 RFC4566 Section 5.2 says that it should make the username, session id,
680 nettype, addrtype and unicast address tuple globally unique. Always using
681 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
682 Instead let's create a 64 bit random number, which at least brings us
683 closer to the goal of global uniqueness.
684 https://tools.ietf.org/html/rfc4566#section-5.2
686 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
688 * examples/test-launch.c:
689 * examples/test-mp4.c:
690 * examples/test-ogg.c:
691 * examples/test-uri.c:
692 examples: Don't call gst_init() and gst_get_option_group()
693 The latter calls the former at the appropriate time.
695 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
697 * gst/rtsp-server/rtsp-client.c:
698 rtsp-client: Drop trailing \0 of RTSP DATA messages
699 We add a trailing \0 in GstRTSPConnection to make parsing of
700 string message bodies easier (e.g. the SDP from DESCRIBE) but
701 for actual data this means we have to drop it or otherwise
704 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
706 * gst/rtsp-server/rtsp-stream.c:
707 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
708 Fixes crash when two threads access handle_new_sample() at the same
709 time, one for RTP, one for RTCP.
710 Otherwise, when iterating over the transports cache, it might be modified by
711 another thread at the same time if the transports cookie has changed.
712 https://bugzilla.gnome.org/show_bug.cgi?id=742954
714 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
716 * gst/rtsp-server/rtsp-stream.c:
717 rtsp-stream: Set format=TIME on our app sources for TCP
719 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
721 * gst/rtsp-server/rtsp-session-pool.c:
722 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
723 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
724 RFC 2326 states that session IDs may consist of alphanumeric as well as
725 the safe characters $-_.+ -- N.B. the percent character is not allowed.
726 Previously the session ID was URI-escaped, this meant that any character
727 which was not alphanumeric or any of the characters +-._~ would be
728 percent encoded. While the RFC (surprisingly) mentions that linear white
729 space in session IDs should be URI-escaped, it does not say anything
730 about other characters. Moreover no white space is allowed in the
731 session ID. Finally the percent character which is the result of
732 URI-escaping is not allowed in a session ID.
733 So there is no reason to do any URI-escaping, and now it is removed.
734 https://bugzilla.gnome.org/show_bug.cgi?id=742869
736 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
739 Automatic update of common submodule
740 From f2c6b95 to bc76a8b
742 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
745 Fix 'make check' from top-level directory
747 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
749 * examples/test-launch.c:
750 * examples/test-mp4.c:
751 * examples/test-ogg.c:
752 * examples/test-uri.c:
753 examples: Add command-line parsing and take a 'port' argument
754 This allows users to run multiple servers on different ports for testing.
755 Only done for examples that actually take arguments and hence are capable of
756 outputting different streams for each instance on each port.
757 https://bugzilla.gnome.org/show_bug.cgi?id=742115
759 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
761 * gst/rtsp-server/rtsp-client.c:
762 * gst/rtsp-server/rtsp-client.h:
763 rtsp-client: Add a send_message default signal handler
764 This allows subclasses to easily hook into the response sending
765 mechanism without doing everything from a signal, which seems
766 awkward from subclasses.
768 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
771 Automatic update of common submodule
772 From ef1ffdc to f2c6b95
774 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
778 configure: add --disable-examples switch
779 https://bugzilla.gnome.org/show_bug.cgi?id=741678
781 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
783 * examples/.gitignore:
784 * examples/Makefile.am:
785 * examples/test-video-rtx.c:
786 examples: add a retransmisison example implementing RFC4588
787 Currently only SSRC-multiplexed rtx streams are supported
789 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
791 * gst/rtsp-server/rtsp-stream.c:
792 rtsp-stream: Fix some minor memory leaks
794 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
796 * gst/rtsp-server/rtsp-media.c:
797 rtsp-media: Some minor cleanup
799 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
801 * gst/rtsp-server/rtsp-stream.c:
802 rtsp-stream: Fix compiler warnings
803 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
804 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
806 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
807 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
810 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
812 * docs/libs/gst-rtsp-server-sections.txt:
813 * gst/rtsp-server/rtsp-media-factory.c:
814 * gst/rtsp-server/rtsp-media-factory.h:
815 * gst/rtsp-server/rtsp-media.c:
816 * gst/rtsp-server/rtsp-media.h:
817 * gst/rtsp-server/rtsp-sdp.c:
818 * gst/rtsp-server/rtsp-stream.c:
819 * gst/rtsp-server/rtsp-stream.h:
820 media: implement ssrc-multiplexed retransmission support
821 based off RFC 4588 and the server-rtpaux example in -good
823 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
825 * gst/rtsp-server/rtsp-client.c:
826 * gst/rtsp-server/rtsp-stream-transport.c:
827 * gst/rtsp-server/rtsp-stream.c:
828 rtsp: Ref transports in hash table.
829 Also ref streams for transports.
830 This solves a crash when reciving a rtcp after teardown but before
832 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
834 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
837 Automatic update of common submodule
838 From 7bb2bce to ef1ffdc
840 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
842 * gst/rtsp-server/rtsp-client.c:
843 client: refactor cleanup of cached media
845 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
847 * tests/check/gst/client.c:
849 The session leak is now fixed, lets remove those FIXME comments.
851 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
853 * tests/check/gst/rtspserver.c:
854 tests: Test to setup two sessions on one connection
855 https://bugzilla.gnome.org/show_bug.cgi?id=739112
857 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
859 * tests/check/gst/rtspserver.c:
860 tests: Test setup with tcp transport
861 https://bugzilla.gnome.org/show_bug.cgi?id=739112
863 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
865 * gst/rtsp-server/rtsp-client.c:
866 client: Configure transport after creating session media
867 The default implementation of configure_client_transport() in
868 rtsp-client uses the session media when it chooses channels for
870 https://bugzilla.gnome.org/show_bug.cgi?id=739112
872 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
874 * gst/rtsp-server/rtsp-client.c:
875 * gst/rtsp-server/rtsp-session-media.c:
876 client: Stop caching media in client when doing setup
877 If the media has been managed by a session media, it should not be
878 cached in the client any longer. The GstRTSPSessionMedia object is now
879 responsible for unpreparing the GstRTSPMedia object using
880 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
882 https://bugzilla.gnome.org/show_bug.cgi?id=739112
884 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
886 * gst/rtsp-server/rtsp-stream.c:
887 rtsp-stream: unref srtp decoder when leaving bin
888 https://bugzilla.gnome.org/show_bug.cgi?id=739481
890 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
892 * gst/rtsp-server/rtsp-client.c:
893 rtsp-client: mikey memory leaks
894 https://bugzilla.gnome.org/show_bug.cgi?id=739383
896 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
899 Automatic update of common submodule
900 From 84d06cd to 7bb2bce
902 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
905 Parallelise 'make check-valgrind'
907 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
910 Automatic update of common submodule
911 From a8c8939 to 84d06cd
913 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
916 Automatic update of common submodule
917 From 36388a1 to a8c8939
919 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
921 * gst/rtsp-server/rtsp-media.c:
922 rtsp-media: deactivate media when shutting down from paused
923 This was only done when going directly from playing.
924 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
926 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
928 * gst/rtsp-server/rtsp-client.c:
929 * gst/rtsp-server/rtsp-context.h:
930 rtsp-client: add stream transport to context
931 We add the stream transport to the context so we can get the configured
932 client stream transport in the setup request signal.
933 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
935 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
937 * gst/rtsp-server/rtsp-stream.c:
938 stream: release lock even not all transports have been removed
939 We don't want to keep the lock even we return FALSE because not all the
940 transports have been removed. This could lead into a deadlock.
941 https://bugzilla.gnome.org/show_bug.cgi?id=737797
943 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
945 * gst/rtsp-server/rtsp-sdp.c:
946 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
947 These were renamed in GstRTPBasePayload in 1.0
949 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
951 * gst/rtsp-server/rtsp-client.c:
952 client: set session media to NULL without the lock
953 We need to set session medias to NULL without the client lock otherwise
954 we can end up in a deadlock if another thread is waiting for the lock
955 and media unprepare is also waiting for that thread to end.
956 https://bugzilla.gnome.org/show_bug.cgi?id=737690
958 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
960 * gst/rtsp-server/rtsp-media.c:
961 rtsp-media: Set state to UNPREPARING in all cases
963 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
965 * gst/rtsp-server/rtsp-media.c:
966 media: set state to unpreparing when unprepare is initiated
967 https://bugzilla.gnome.org/show_bug.cgi?id=737675
969 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
971 * gst/rtsp-server/rtsp-client.c:
972 rtsp-client: Remove backlog limit while processings requests
973 If the backlog limit is kept two cases of deadlocks may be
974 encountered when streaming over TCP. Without the backlog
975 limit this deadlocks can not happen, at the expence of
977 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
979 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
981 * gst/rtsp-server/rtsp-client.c:
982 rtsp-client: do not free main context before rtsp watch
983 https://bugzilla.gnome.org/show_bug.cgi?id=737110
985 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
987 * tests/check/gst/rtspserver.c:
988 tests: Extend unit test timeout to accomodate for valgrind
989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
991 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
993 * gst/rtsp-server/rtsp-client.c:
994 * gst/rtsp-server/rtsp-session.c:
995 * gst/rtsp-server/rtsp-stream-transport.c:
996 rtsp-*: Treat sending packets to clients as keepalive
997 As long as gst-rtsp-server can successfully send RTP/RTCP data to
998 clients then the client must be reading. This change makes the server
999 timeout the connection if the client stops reading.
1000 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1002 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1004 * gst/rtsp-server/rtsp-client.c:
1005 rtsp-client: Allow backlog to grow while expiring session
1006 Allow the send backlog in the RTSP watch to grow to unlimited size while
1007 attempting to bring the media pipeline to NULL due to a session
1008 expiring. Without this change the appsink element cannot change state
1009 because it is blocked while rendering data in the new_sample callback.
1010 This callback will block until it has successfully put the data into the
1011 send backlog. There is a chance that the send backlog is full at this
1012 point which means that the callback may block for a long time, possibly
1013 forever. Therefore the media pipeline may also be prevented from
1014 changing state for a long time.
1015 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1017 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1019 * gst/rtsp-server/rtsp-client.c:
1020 rtsp-client: Make old compilers happy
1021 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1022 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1024 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1026 * gst/rtsp-server/rtsp-client.c:
1027 client: raise the backlog limits before pausing
1028 We need to raise the backlog limits before pausing the pipeline or else
1029 the appsink might be blocking in the render method in wait_backlog() and
1030 we would deadlock waiting for paused.
1031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1033 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1035 * gst/rtsp-server/rtsp-client.c:
1036 client: make define for the WATCH_BACKLOG
1037 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1039 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1041 * gst/rtsp-server/rtsp-client.c:
1042 client: simplify session transport handling
1043 link/unlink of the transport in a session was done to keep track of all
1044 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1045 that by putting all the TCP transports in a hashtable indexed with the
1047 We also don't need to link/unlink the transports when we pause/resume
1048 the streams. The same effect is already achieved when we pause/play the
1049 media. Indeed, when we pause the media, the transport is removed from
1050 the media and the callbacks will not be called anymore.
1051 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1053 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1055 * gst/rtsp-server/rtsp-stream-transport.c:
1056 * gst/rtsp-server/rtsp-stream-transport.h:
1057 stream-transport: make method to handle received data
1058 Make a method to handle the data received on a channel. It sends the
1059 data to the stream of the transport on the RTP or RTCP pads based on
1062 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1064 * examples/test-mp4.c:
1065 test: add example of dumping RTCP reports
1067 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1069 * gst/rtsp-server/rtsp-media.c:
1070 * gst/rtsp-server/rtsp-stream.c:
1071 * gst/rtsp-server/rtsp-stream.h:
1072 rtsp-media: Make sure that sequence numbers are monotonic after pause
1073 The sequence number is not monotonic for RTP packets after pause. The
1074 reason is basepayloader generates a randon sequence number when the
1075 pipeline goes from ready to pause. With this fix generation of sequence
1076 number will be monotonic when going from pause to play request.
1077 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1079 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1081 * gst/rtsp-server/rtsp-client.c:
1082 rtsp-client: Protect saved clients watch with a mutex
1083 Fixes a crash when close() is called while merging clients
1084 in handle_tunnel(). In that case close() would destroy the
1085 watch while it is still being used in handle_tunnel().
1086 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1088 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1090 * gst/rtsp-server/rtsp-stream.c:
1091 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1093 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1095 * gst/rtsp-server/rtsp-media.c:
1096 * gst/rtsp-server/rtsp-stream.c:
1097 * gst/rtsp-server/rtsp-stream.h:
1098 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1099 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1100 seeking and will always continue counting the time. This leads to
1101 the NPT after a backwards seek to be something completely different
1102 to the actual seek position.
1103 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1105 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1107 * examples/test-appsrc.c:
1108 examples: fix another reference leak
1109 gst_rtsp_media_get_element() returns a new ref.
1111 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1113 * examples/test-appsrc.c:
1114 examples: unref element after usage
1115 gst_bin_get_by_name_recurse_up() returns an element
1116 reference that must be unreffed after usage.
1117 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1119 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1121 * gst/rtsp-server/rtsp-media.c:
1122 signals: Fix copy-pasto in target-state signal offset
1124 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1128 Makefile: Add usage of build-checks step
1129 Allows building checks without running them
1131 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1133 * gst/rtsp-server/rtsp-stream.c:
1134 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1135 When a UDP multicast transport is used it is expected that the server listens
1136 for RTP and RTCP packets on the multicast group with the corresponding port.
1137 Without this we will never get RTCP packets from clients in multicast mode.
1138 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1140 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1145 === release 1.4.0 ===
1147 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1153 * gst-rtsp-server.doap:
1156 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1158 * gst/rtsp-server/rtsp-media.h:
1159 media: correct misspelled words in description
1160 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1162 === release 1.3.91 ===
1164 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1170 * gst-rtsp-server.doap:
1173 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1175 * docs/libs/gst-rtsp-server-sections.txt:
1178 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1180 * gst/rtsp-server/rtsp-server.c:
1181 server: implement client REMOVE filter
1183 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1185 * gst/rtsp-server/rtsp-client.c:
1186 * gst/rtsp-server/rtsp-client.h:
1187 client: expose _close() method
1188 Expose a previously internal close method to close the client
1191 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1193 * gst/rtsp-server/rtsp-session-pool.c:
1194 session-pool: signal session-removed outside of the lock
1195 Release the lock before emiting the session-removed signal.
1197 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1199 * gst/rtsp-server/rtsp-client.c:
1200 * gst/rtsp-server/rtsp-server.c:
1201 * gst/rtsp-server/rtsp-session-pool.c:
1202 * gst/rtsp-server/rtsp-session.c:
1203 * gst/rtsp-server/rtsp-stream.c:
1204 filter: Release lock in filter functions
1205 Release the object lock before calling the filter functions. We need to
1206 keep a cookie to detect when the list changed during the filter
1207 callback. We also keep a hashtable to make sure we only call the filter
1208 function once for each object in case of concurrent modification.
1209 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1211 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1213 * gst/rtsp-server/rtsp-client.c:
1214 client: check if watch is set in handle_teardown()
1215 The unit tests run without a watch
1217 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1219 * tests/check/gst/client.c:
1220 client tests: send teardown to cleanup session
1222 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1224 * tests/check/gst/rtspserver.c:
1225 server tests: send teardown to cleanup session
1227 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1229 * gst/rtsp-server/rtsp-client.c:
1230 client: keep ref to client for the session removed handler
1231 This extra ref will be dropped when all client sessions have been
1232 removed. A session is removed when a client sends teardown, closes its
1233 endpoint of the TCP connection or the sessions expires.
1234 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1236 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1238 * gst/rtsp-server/rtsp-client.c:
1239 * gst/rtsp-server/rtsp-session.c:
1240 * tests/check/gst/client.c:
1241 client: manage media in session as a last step
1242 Once we manage a media in a session, we can't unmanage it anymore
1243 without destroying it. Therefore, first check everything before we
1244 manage the media, otherwise if something is wrong we have no way to
1246 If we created a new session and something went wrong, remove the session
1247 again. Fixes a leak in the unit test.
1249 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1251 * examples/test-mp4.c:
1252 * examples/test-ogg.c:
1253 examples: print 'stream ready at url' for mp4 and ogg example
1255 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1257 * gst/rtsp-server/rtsp-client.c:
1258 * gst/rtsp-server/rtsp-sdp.c:
1259 rtsp: fix for MIKEY api change
1261 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1263 * gst/rtsp-server/rtsp-client.c:
1264 client: free watch context only once
1265 The watch context is freed when the source is destroyed. Avoids
1266 a CRITICAL when we try to unref the context twice.
1268 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1270 * gst/rtsp-server/rtsp-client.c:
1273 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1275 * gst/rtsp-server/rtsp-client.c:
1276 client: protect sessions with lock
1277 Protect the list of sessions with the lock.
1278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1280 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1282 * gst/rtsp-server/rtsp-client.c:
1283 Client: keep a ref to the session
1284 Don't just keep a weak ref to the session objects but use a hard ref. We
1285 will be notified when a session is removed from the pool (expired) with
1286 the new session-removed signal.
1287 Don't automatically close the RTSP connection when all the sessions of
1288 a client are removed, a client can continue to operate and it can create
1289 a new session if it wants. If you want to remove the client from the
1290 server, you have to use gst_rtsp_server_client_filter() now.
1291 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1292 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1294 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1296 * gst/rtsp-server/rtsp-session-pool.c:
1297 * gst/rtsp-server/rtsp-session-pool.h:
1298 session-pool: add session-removed signal
1299 Add a signal to be notified when a session is removed from the pool.
1301 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1303 * gst/rtsp-server/Makefile.am:
1304 * gst/rtsp-server/rtsp-server.h:
1305 Make rtsp-server.h a single-include header, use it for G-I
1306 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1308 === release 1.3.90 ===
1310 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1316 * gst-rtsp-server.doap:
1319 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1321 * gst/rtsp-server/rtsp-stream.c:
1322 stream: crypto can be NULL
1324 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1326 * gst/rtsp-server/rtsp-client.c:
1327 * gst/rtsp-server/rtsp-media.c:
1328 * gst/rtsp-server/rtsp-mount-points.c:
1329 introspection: add missing allow-none annotations
1330 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1332 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1334 * gst/rtsp-server/rtsp-address-pool.c:
1335 * gst/rtsp-server/rtsp-media.c:
1336 * gst/rtsp-server/rtsp-session-media.c:
1337 * gst/rtsp-server/rtsp-session-pool.c:
1338 * gst/rtsp-server/rtsp-stream-transport.c:
1339 * gst/rtsp-server/rtsp-stream.c:
1340 * gst/rtsp-server/rtsp-token.c:
1341 introspection: add (nullable) annotations to return values
1342 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1344 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1346 * gst/rtsp-server/rtsp-client.c:
1347 * gst/rtsp-server/rtsp-stream.c:
1348 gi: improve annotations
1349 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1351 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1353 * gst/rtsp-server/rtsp-client.c:
1354 * gst/rtsp-server/rtsp-media-factory.c:
1355 * gst/rtsp-server/rtsp-media.c:
1356 * gst/rtsp-server/rtsp-server.c:
1357 signals: use generic marshal function
1358 Use the generic C marshal function.
1359 Use more explicit type instead of G_TYPE_POINTER
1361 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1363 * gst/rtsp-server/rtsp-context.h:
1364 context: add type macro
1366 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1368 * gst/rtsp-server/rtsp-client.c:
1369 * gst/rtsp-server/rtsp-sdp.c:
1370 * gst/rtsp-server/rtsp-sdp.h:
1371 sdp: hide key length defines
1372 They don't have a namespace.
1374 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1379 === release 1.3.3 ===
1381 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1387 * gst-rtsp-server.doap:
1390 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1392 * gst/rtsp-server/rtsp-client.c:
1393 * gst/rtsp-server/rtsp-sdp.c:
1394 * gst/rtsp-server/rtsp-sdp.h:
1395 mikey: add different key length parameters
1396 Add encryption and authentication key length parameters to MIKEY. For
1397 the encoders, the key lengths are obtained from the cipher and auth
1398 algorithms set in the caps. For the decoders, they are obtained while
1399 parsing the key management from the client.
1400 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1402 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1404 * tests/check/gst/stream.c:
1405 stream tests: Make sure we get right multicast address from stream
1406 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1408 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1410 * gst/rtsp-server/rtsp-client.c:
1411 client: ref the context until rtsp watch is alive
1412 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1414 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1416 * gst/rtsp-server/rtsp-client.c:
1417 client: Destroy the rtsp watch after connection close
1419 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1421 * gst/rtsp-server/rtsp-media.c:
1422 media: fix confusing comment
1424 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1426 * gst/rtsp-server/rtsp-session.c:
1427 rtsp-session: Timeout in header.
1428 Adding the possbilty to always have timout in header.
1429 This is configurabe with setting "timeout-always-visible".
1430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1432 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1437 === release 1.3.2 ===
1439 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1446 * gst-rtsp-server.doap:
1449 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1452 Automatic update of common submodule
1453 From 211fa5f to 1f5d3c3
1455 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1457 * gst/rtsp-server/rtsp-client.c:
1458 client: store TCP ports in transport
1459 Store the TCP ports in the transport when we are doing RTSP over TCP.
1460 This way, we can easily get to the ports from the transport.
1461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1463 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1465 * gst/rtsp-server/rtsp-stream.c:
1466 stream: add signals for new RTP/RTCP encoders
1467 New signals to allow the user to configure the dynamically created
1469 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1471 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1473 * gst/rtsp-server/rtsp-media.c:
1474 * gst/rtsp-server/rtsp-media.h:
1475 media: Make suspend()/unsuspend() virtual
1476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1478 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1480 * gst/rtsp-server/rtsp-client.c:
1481 client: fix send-message signal marshaller
1482 Use generic marshalling for the send-message signal. It has
1483 two POINTER arguments, not just one.
1484 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1486 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1488 * tests/check/gst/media.c:
1489 tests: add and remove pads only once
1490 In this test we simulate a dynamic pad by watching the caps event.
1491 Because of renegotiation in the base payloader now, this caps is sent
1492 multiple times but we can only deal with 1 invocation, use a variable to
1493 only 'add and remove' the pad once.
1495 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1497 * tests/check/gst/rtspserver.c:
1498 tests: add unit test for correct handling of Require headers
1499 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1501 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1503 * gst/rtsp-server/rtsp-client.c:
1504 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1505 Servers must handle Require headers and must report a failure
1506 if they don't handle any of the Required options, see RFC 2326,
1507 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1508 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1510 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1515 === release 1.3.1 ===
1517 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1523 * gst-rtsp-server.doap:
1526 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1529 Automatic update of common submodule
1530 From bcb1518 to 211fa5f
1532 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1537 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1539 * tests/check/gst/sessionmedia.c:
1540 tests: fix memory leak in sessionmedia unit test
1542 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1544 * gst/rtsp-server/rtsp-client.c:
1545 client: emit a signal before sending a message
1546 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1548 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1550 * gst/rtsp-server/rtsp-client.c:
1551 client: pass context to send_message
1552 Pass the current context to send_message, we will need it later.
1554 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1556 * gst/rtsp-server/rtsp-client.c:
1557 client: fix typo in comment
1559 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1561 * gst/rtsp-server/rtsp-media.c:
1562 media: Do not stop thread twice if default_prepare() fails
1564 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1566 * gst/rtsp-server/rtsp-client.c:
1567 client: set the watch to flushing before going to NULL
1568 First set the watch to flushing so that we unblock any current and
1569 future attempt to send data on the watch, Then set the pipeline to
1571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1573 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1575 * gst/rtsp-server/rtsp-session-pool.c:
1576 * tests/check/gst/sessionpool.c:
1577 rtsp-session-pool: Fixes annotation
1578 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1579 in the sessionpool test.
1580 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1582 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1584 * gst/rtsp-server/rtsp-media.c:
1585 * gst/rtsp-server/rtsp-media.h:
1586 media: make media_prepare virtual
1587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1589 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1591 * gst/rtsp-server/rtsp-media.c:
1592 * tests/check/gst/media.c:
1593 media: stop the thread in more error cases
1595 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1597 * gst/rtsp-server/rtsp-media.c:
1598 * tests/check/gst/media.c:
1599 media: allow NULL as the thread
1600 Use the default context whan passing a NULL thread.
1602 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1604 * gst/rtsp-server/rtsp-client.c:
1605 rtsp-client: indent cleanup
1606 Coverity was moaning about unreachable code, and I think it was just
1607 confused by { being before the label. We'll see if it pops up again.
1610 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1612 * gst/rtsp-server/rtsp-client.c:
1613 * gst/rtsp-server/rtsp-media.c:
1614 client: Add drop-backlog property
1615 When we have too many messages queued for a client (currently hardcoded
1616 to 100) we overflow and drop the messages. Add a drop-backlog property
1617 to control this behaviour. Setting this property to FALSE will retry
1618 to send the messages to the client by waiting for more room in the
1620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1622 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1624 * gst/rtsp-server/rtsp-client.c:
1625 client: support for POST before GET when setting up a tunnel
1627 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1629 * gst/rtsp-server/rtsp-client.c:
1630 client: remove watch of the second client after http tunnel setup
1631 The second client will be freed after the HTTP tunnel has been set up.
1632 Make sure it's RTSP watch is never dispatched again.
1633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1635 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1637 * gst/rtsp-server/rtsp-media.c:
1638 * tests/check/gst/media.c:
1639 media: Make media_prepare() fail if port allocation fails
1640 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1642 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1644 * tests/check/gst/media.c:
1645 media test: cleanup the thread pool in tests
1647 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1649 * gst/rtsp-server/rtsp-media.c:
1650 * tests/check/gst/media.c:
1651 rtsp-media: Unblock blocked streams in unprepare
1652 The streams will be blocked when a live media is prepared.
1653 The streams should be unblocked in gst_rtsp_media_unprepare.
1654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1656 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1658 * gst/rtsp-server/rtsp-media.c:
1659 media: release the state lock when going to NULL
1660 Set our state to UNPREPARING and release the state-lock before
1661 setting the pipeline to the NULL state. This way, any pad-added
1662 callback will be able to take the state-lock and check that we are now
1663 unpreparing instead of deadlocking.
1664 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1666 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1668 * gst/rtsp-server/rtsp-media.c:
1669 media: protect status with lock
1670 Make sure we only update the status with the lock.
1672 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1674 * gst/rtsp-server/rtsp-client.c:
1675 * gst/rtsp-server/rtsp-sdp.c:
1676 rtsp: update for MIKEY API changes
1678 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1680 * gst/rtsp-server/rtsp-client.c:
1681 client: parse the mikey response from the client
1682 Parse the mikey response from the client and update the policy for
1685 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1687 * gst/rtsp-server/rtsp-stream.c:
1688 * gst/rtsp-server/rtsp-stream.h:
1689 stream: add method to set crypto info
1690 Make a method to configure the crypto information of a stream.
1691 Set udpsrc in READY instead of PAUSED so that we can configure caps
1694 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1696 * gst/rtsp-server/rtsp-client.c:
1697 client: cleanup error paths
1699 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1701 * gst/rtsp-server/rtsp-media.c:
1704 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1706 * examples/test-video.c:
1707 test: enable SRTP only on RTSPS
1708 We only want to enable SRTP when doing rtsp over TLS so that we can
1709 exchange the keys in a secure way.
1711 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1713 * examples/test-video.c:
1714 test: print an error on failure
1716 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1719 * examples/test-video.c:
1720 * gst/rtsp-server/rtsp-sdp.c:
1721 * gst/rtsp-server/rtsp-stream.c:
1722 * tests/check/Makefile.am:
1723 stream: add SRTP support
1724 Install srtp encoder and decoder elements in rtpbin
1727 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1729 * tests/check/Makefile.am:
1730 * tests/check/gst/sessionpool.c:
1731 tests: Add unit tests for sessionpool
1732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1734 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1736 * tests/check/gst/threadpool.c:
1737 tests: Improve code coverage of rtsp-threadpool tests
1738 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1740 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1742 * tests/check/gst/sessionmedia.c:
1743 tests: Improve code coverage for rtsp-session-media
1744 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1746 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1748 gobject-introspection: Add annotations to support language bindings
1749 In addition a few cosmetic changes:
1750 * Adjust the order of arguments
1751 * Fix typo: occured -> occurred
1752 * Fix indentation after Return:-clauses
1753 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1755 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1757 * gst/rtsp-server/rtsp-stream.c:
1758 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1759 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1761 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1763 * gst/rtsp-server/rtsp-stream.c:
1764 stream: take caps after the session manager
1765 Take the caps for the SDP after they leave the rtpbin so that we can
1766 also get the properties added by rtpbin elements.
1768 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1770 * gst/rtsp-server/rtsp-stream.c:
1771 stream: release lock while pushing out packets
1772 Keep a cache of the transports and use this to iterate the transport
1773 while pushing packets. This allows us to release the lock early.
1774 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1776 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1778 * gst/rtsp-server/rtsp-client.c:
1779 * gst/rtsp-server/rtsp-client.h:
1780 rtsp-client: vmethod for modifying tunnel GET response
1781 Add a vmethod tunnel_http_response where the response to the HTTP GET
1782 for tunneled connections can be modified.
1783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1785 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1787 * gst/rtsp-server/rtsp-sdp.c:
1788 sdp: make 1 media line per profile
1789 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1790 line in the SDP for each profile. The client is then supposed to pick
1791 one of the profiles in the SETUP request. Because the m= lines have the
1792 same pt, the client also knows that only 1 option is possible.
1794 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1796 * gst/rtsp-server/rtsp-media-factory.c:
1797 * gst/rtsp-server/rtsp-media-factory.h:
1798 * gst/rtsp-server/rtsp-media.c:
1799 factory: add profile property and pass to media and streams
1801 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1803 * examples/test-multicast.c:
1804 * gst/rtsp-server/rtsp-sdp.c:
1805 sdp: pass multicast connection for multicast-only stream
1806 Pass the multicast address of the stream in the connection info in the
1807 SDP so that clients try a multicast connection first.
1808 Only allow multicast connections in the test-multicast example. Also
1809 increase the TTL a little.
1811 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1814 .gitignore: Ignore gcov intermediate files
1815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1817 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1819 * gst/rtsp-server/rtsp-stream.c:
1820 stream: release some locks in error cases
1822 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1824 docs: Enable and fix gtk-doc warnings
1825 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1826 * addresspool/mediafactory: Add missing annotation colon
1827 * stream: Annotate return value
1828 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1830 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1833 Automatic update of common submodule
1834 From fe1672e to bcb1518
1836 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1839 Automatic update of common submodule
1840 From 1a07da9 to fe1672e
1842 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1844 * examples/Makefile.am:
1845 examples: use LDADD for libs instead of LDFLAGS
1847 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1850 configure: make sure releases are in .doap file
1852 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1854 * examples/test-cgroups.c:
1855 examples: test-cgroups: don't put code with side effects into g_assert()
1856 The g_assert() might get compiled out with the right
1857 compiler/preprocessor flags.
1859 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1861 * examples/.gitignore:
1862 examples: add cgroup test binary to .gitignore
1864 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1866 * examples/test-cgroups.c:
1867 examples: fix cgroup test build
1868 Fixes build failure caused by compiler warning:
1869 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1871 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1874 .gitignore: ignore temp files created in the course of 'make check'
1876 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1878 * gst/rtsp-server/rtsp-media.c:
1879 rtsp-media: don't loose frames handling new PLAY request
1880 If client supplied a range check if the range specifies the start point.
1881 If not, then do an accurate seek to the current position. If a start
1882 point was specified do do a key unit seek to make sure the streaming
1883 starts with decodeable frames.
1884 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1886 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1888 * gst/rtsp-server/rtsp-media.c:
1889 Revert "media: only flush when setting a new start position"
1890 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1891 We need to do the flush in all cases, demuxer block currently for
1894 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1896 * gst/rtsp-server/rtsp-media.c:
1897 media: only flush when setting a new start position
1898 Only flush the pipeline when we change the start position with
1900 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
1902 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
1904 * gst/rtsp-server/rtsp-stream.c:
1905 stream: set ttl-mc before adding the socket
1906 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
1907 never be set on socket.
1908 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
1910 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1912 * gst/rtsp-server/rtsp-media.c:
1913 media: stop thread if media is already prepared
1914 in gst_rtsp_media_prepare() the thread is not used if media is already
1915 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
1917 https://bugzilla.gnome.org/show_bug.cgi?id=724182
1919 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
1922 build: Ship gst-rtsp-server.doap file
1924 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
1926 * tests/check/gst/rtspserver.c:
1927 tests: Fix another compiler warning with gcc
1929 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
1931 * gst/rtsp-server/rtsp-client.c:
1932 * gst/rtsp-server/rtsp-mount-points.c:
1933 * gst/rtsp-server/rtsp-stream.c:
1934 * tests/check/gst/client.c:
1935 rtsp-server: Fix lots of compiler warnings with clang
1937 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
1940 * gst-rtsp-server.doap:
1941 * tests/Makefile.am:
1942 configure: Synchronise with the configure scripts of the other modules
1944 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1947 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
1949 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1951 * gst/rtsp-server/rtsp-media.c:
1952 * gst/rtsp-server/rtsp-stream.c:
1953 Revert "rtsp-server: support build against last stable release"
1954 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
1955 Let us require 1.2.3 now, which is going to be released in a few
1958 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
1960 * gst/rtsp-server/rtsp-session-media.c:
1961 * gst/rtsp-server/rtsp-stream-transport.c:
1962 session: improve RTP-Info
1963 Ignore streams that can't generate RTP-Info instead of failing.
1964 Don't return the empty string when all streams are unconfigured but
1965 return NULL so that we don't generate and empty RTP-Info header.
1966 Improve docs a little.
1968 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
1970 * gst/rtsp-server/rtsp-session-media.c:
1971 Don't free rtpinfo GString when it is NULL
1972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
1974 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
1976 * gst/rtsp-server/rtsp-media.c:
1977 media: only set keyframe flag when modifying start
1978 Only set the keyframe flag when we modify the start position. The
1979 keyframe flag should probably be ignored when no change is requested but
1980 until we can claim this is all documented properly and all demuxer
1981 implement this, avoid setting the flag.
1982 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
1984 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
1986 * gst/rtsp-server/rtsp-thread-pool.c:
1987 thread-pool: Unref source after mainloop has quit to avoid races in GLib
1988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
1990 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
1992 * gst/rtsp-server/rtsp-stream.c:
1993 stream: handle NULL seqnum and rtptime arguments
1995 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
1997 * gst/rtsp-server/rtsp-thread-pool.c:
1998 * tests/check/gst/threadpool.c:
1999 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2000 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2002 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2004 * gst/rtsp-server/rtsp-stream.c:
2005 stream: add fallback for missing stats property
2006 Use a fallback when the payloader does not have a stats property
2007 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2009 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2012 Automatic update of common submodule
2013 From f7bc1c3 to 1a07da9
2015 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2017 * gst/rtsp-server/rtsp-stream.c:
2018 stream: don't leak stats structure
2019 Don't leak the stats structure and deal with NULL stats.
2021 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2023 * gst/rtsp-server/rtsp-stream.c:
2024 stream: Get rtpinfo properties atomically from payloader
2025 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2027 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2029 * gst/rtsp-server/rtsp-media.c:
2030 media: refactor state change functions and signals
2031 Make functions to set the target state and the pipeline state and emit
2032 the signals from those functions.
2034 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2036 * gst/rtsp-server/rtsp-media.c:
2037 * gst/rtsp-server/rtsp-media.h:
2038 media: add signal to notify of pending state changes
2040 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2042 * gst/rtsp-server/rtsp-media.c:
2043 * gst/rtsp-server/rtsp-stream.c:
2044 rtsp-server: support build against last stable release
2045 Until 1.2.3 is out with the new get_type function and we
2048 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2050 * gst/rtsp-server/rtsp-stream.c:
2051 stream: fix compilation
2053 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2055 * gst/rtsp-server/rtsp-media.c:
2056 * gst/rtsp-server/rtsp-media.h:
2057 * gst/rtsp-server/rtsp-stream.c:
2058 * gst/rtsp-server/rtsp-stream.h:
2059 stream: add property to configure profiles
2061 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2063 * gst/rtsp-server/rtsp-client.c:
2064 client: let stream check supported transport
2065 Delegate the check if a transport is allowed to the stream.
2066 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2068 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2070 * gst/rtsp-server/rtsp-stream.c:
2071 * gst/rtsp-server/rtsp-stream.h:
2072 stream: add method to check supported transport
2073 Add a method to check if a transport is supported
2075 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2078 configure.ac: Only check for gstreamer-check, not check
2079 We include check in gstreamer-check since quite some time now.
2081 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2083 * gst/rtsp-server/rtsp-session-media.c:
2084 * gst/rtsp-server/rtsp-stream-transport.c:
2085 * gst/rtsp-server/rtsp-stream.c:
2086 * gst/rtsp-server/rtsp-stream.h:
2087 stream: return clock-rate from get_rtpinfo
2088 And use it to correct the rtptime to the requested start-time.
2089 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2091 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2093 * gst/rtsp-server/rtsp-session-media.c:
2094 * gst/rtsp-server/rtsp-stream-transport.c:
2095 * gst/rtsp-server/rtsp-stream-transport.h:
2096 session-media: calculate start-time
2098 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2100 * gst/rtsp-server/rtsp-stream-transport.c:
2101 * gst/rtsp-server/rtsp-stream.c:
2102 * gst/rtsp-server/rtsp-stream.h:
2103 stream: also return the running-time
2104 Return the running-time in the rtpinfo as well.
2106 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2108 * gst/rtsp-server/rtsp-client.c:
2109 * gst/rtsp-server/rtsp-session-media.c:
2110 * gst/rtsp-server/rtsp-session-media.h:
2111 * gst/rtsp-server/rtsp-stream-transport.c:
2112 * gst/rtsp-server/rtsp-stream-transport.h:
2113 session-media: let the session-media make the RTPInfo
2114 Add method to create the RTPInfo for a stream-transport.
2115 Add method to create the RTPInfo for all stream-transports in a
2117 Use the session-media RTPInfo code in client. This allows us to refactor
2118 another method to link the TCP callbacks.
2120 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2122 mount-points: sort sequence before g_sequence_lookup
2123 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2124 sort sequence if dirty, otherwise lookup will fail.
2125 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2127 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2130 configure: rename package from gst-rtsp to gst-rtsp-server
2131 To match git module name and avoid confusion with the
2132 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2134 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2137 configure: bump core/base/good requirement to 1.2.0
2138 Bump to released stable version and make implicit
2139 requirements explicit.
2141 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2146 Fix broken gettext setup which is not used anyway
2148 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2151 Automatic update of common submodule
2152 From dbedaa0 to d48bed3
2154 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2156 * gst/rtsp-server/rtsp-client.c:
2157 * gst/rtsp-server/rtsp-media.c:
2158 * gst/rtsp-server/rtsp-media.h:
2159 media: add setup_sdp vmethod
2160 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2161 gst_rtsp_media_setup_sdp.
2162 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2164 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2166 * gst/rtsp-server/rtsp-stream.c:
2167 rtsp-stream: Check return value of sscanf
2168 streamid is only valid if sscanf matched something.
2170 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2172 * gst/rtsp-server/rtsp-client.c:
2173 rtsp-client: Fix iteration
2174 Wouldn't even enter the code block otherwise (i++ was used as the check
2175 and not the postfix).
2177 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2179 * gst/rtsp-server/rtsp-client.c:
2180 * gst/rtsp-server/rtsp-client.h:
2181 client: add vmethod to configure media and streams
2182 Implement a vmethod that can be used to configure the media and the
2183 streams based on the current context. Handle the blocksize handling in
2184 the default handler.
2185 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2187 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2190 Make git ignore more unit test binaries
2192 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2194 * gst/rtsp-server/rtsp-address-pool.h:
2195 * gst/rtsp-server/rtsp-auth.h:
2196 * gst/rtsp-server/rtsp-client.h:
2197 * gst/rtsp-server/rtsp-context.h:
2198 * gst/rtsp-server/rtsp-media-factory-uri.h:
2199 * gst/rtsp-server/rtsp-media-factory.h:
2200 * gst/rtsp-server/rtsp-media.h:
2201 * gst/rtsp-server/rtsp-mount-points.h:
2202 * gst/rtsp-server/rtsp-server.h:
2203 * gst/rtsp-server/rtsp-session-media.h:
2204 * gst/rtsp-server/rtsp-session-pool.h:
2205 * gst/rtsp-server/rtsp-session.h:
2206 * gst/rtsp-server/rtsp-stream-transport.h:
2207 * gst/rtsp-server/rtsp-stream.h:
2208 * gst/rtsp-server/rtsp-thread-pool.h:
2209 * gst/rtsp-server/rtsp-token.h:
2210 rtsp-server: add padding to many public structures
2211 Not mini objects though, since they are not subclassable
2212 anyway, nor kept on the stack or inlined in a structure.
2214 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2216 media: add new create_rtpbin vmethod
2217 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2218 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2220 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2222 * tests/check/gst/media.c:
2223 tests: fix memory leak, free test's thread pool
2224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2226 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2228 * gst/rtsp-server/rtsp-stream-transport.c:
2229 stream-transport: free url in finalize
2231 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2233 * gst/rtsp-server/rtsp-media.c:
2234 media: also do state change in suspended state
2236 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2238 * gst/rtsp-server/rtsp-client.c:
2239 * gst/rtsp-server/rtsp-media.c:
2240 media: also handle prepare and range in suspended state
2241 When we are suspended, we are already prepared.
2242 We can get the range in the suspended state.
2244 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2246 * tests/check/Makefile.am:
2247 * tests/check/gst/sessionmedia.c:
2248 check: add test for uri in setup
2249 Added unit tests for the new functionality in GstRTSPStreamTransport.
2250 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2252 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2254 * gst/rtsp-server/rtsp-client.c:
2255 client: store setup uri and use in PLAY response
2256 Store the uri used when doing the setup and use that in the PLAY
2258 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2260 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2262 * gst/rtsp-server/rtsp-stream-transport.c:
2263 * gst/rtsp-server/rtsp-stream-transport.h:
2264 stream-transport: add method to get/set url
2266 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2268 * gst/rtsp-server/rtsp-client.c:
2269 client: suspend after SDP and unsuspend before PLAYING
2270 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2271 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2273 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2275 * gst/rtsp-server/rtsp-media-factory.c:
2276 * gst/rtsp-server/rtsp-media-factory.h:
2277 * gst/rtsp-server/rtsp-media.c:
2278 * gst/rtsp-server/rtsp-media.h:
2279 * gst/rtsp-server/rtsp-session-media.c:
2280 * gst/rtsp-server/rtsp-session.c:
2281 * tests/check/gst/media.c:
2282 * tests/check/gst/mediafactory.c:
2283 media: add suspend modes
2284 Add support for different suspend modes. The stream is suspended right after
2285 producing the SDP and after PAUSE. Different suspend modes are available that
2286 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2287 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2288 state and RESET will bring the pipeline to the NULL state.
2289 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2290 this means that the pipeline needs to be prerolled again.
2291 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2292 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2294 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2296 * gst/rtsp-server/rtsp-media.c:
2297 media: start live streams in blocked state
2298 Start live streams in the blocked state and make them preroll using the
2299 messages. This ensure that no data is played by the sink until we explicitly
2300 unblock the stream right before going to PLAYING.
2301 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2303 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2305 * gst/rtsp-server/rtsp-media.c:
2306 media: refactor starting and waiting for preroll
2307 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2308 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2310 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2312 * gst/rtsp-server/rtsp-stream.c:
2313 * gst/rtsp-server/rtsp-stream.h:
2314 stream: add API to block streams
2315 Add an API to block on the streams and make it post a message.
2316 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2317 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2319 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2321 * docs/libs/Makefile.am:
2322 docs: Specify the override file
2323 Even if it's empty (for now) it avoids make distcheck complaining
2325 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2327 * gst/rtsp-server/rtsp-media.c:
2328 media: move default implementations to where they are used
2330 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2332 * gst/rtsp-server/rtsp-media.c:
2333 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2334 We need to take the state_lock when calling this method.
2336 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2338 * gst/rtsp-server/rtsp-media.c:
2339 media: handle add-added on non-bins too
2340 Handle dynamic payloaders that are not bins, as used in the unit-test.
2342 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2344 * gst/rtsp-server/rtsp-media-factory.c:
2345 * gst/rtsp-server/rtsp-media-factory.h:
2346 * gst/rtsp-server/rtsp-media.c:
2347 rtsp-media/-factory: Fix request pad name comments
2348 These must be escaped for gtk-doc to parse the comments without warnings.
2350 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2352 rtsp-media: remove transports if media is in error status
2353 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2354 trying to change to GST_STATE_NULL and media is in error status, we
2355 remove all transports.
2356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2358 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2360 * gst/rtsp-server/rtsp-media.c:
2361 rtsp-media: use element metadata to find payloader
2362 Use the element metadata to find the payloader instead of checking
2364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2366 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2368 rtsp-stream: add getter for payload type
2369 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2370 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2371 element and create the stream with this one instead of the dynpay%d
2373 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2375 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2377 * gst/rtsp-server/rtsp-client.c:
2378 * gst/rtsp-server/rtsp-context.h:
2379 * gst/rtsp-server/rtsp-media.c:
2380 * gst/rtsp-server/rtsp-mount-points.c:
2381 * gst/rtsp-server/rtsp-server.c:
2382 * gst/rtsp-server/rtsp-token.c:
2383 rtsp-*: Refer to NULL as a constant in comments
2385 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2387 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2389 rtsp-*: Fix type name typos in comments
2390 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2391 * rtsp-auth: Refer to part of constant name as text
2392 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2393 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2394 * rtsp-stream: Fix typo when refering to GstBin
2395 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2397 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2400 * docs/libs/gst-rtsp-server-docs.sgml:
2401 * docs/libs/gst-rtsp-server-sections.txt:
2402 docs: Improve documentation
2403 * Include annotation-glossary to quiet gtk-doc
2404 * Rename remaining ClientState -> Context
2405 * Rename object hierarchy file
2406 * Remove stale chapter references
2407 * Add missing function and object references
2408 * Include missing GstRTSPAddressPoolResult
2409 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2411 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2413 * gst/rtsp-server/rtsp-client.c:
2414 * gst/rtsp-server/rtsp-server.c:
2415 * gst/rtsp-server/rtsp-session-pool.c:
2416 * gst/rtsp-server/rtsp-session.c:
2417 * gst/rtsp-server/rtsp-stream.c:
2418 rtsp-server: sprinkle some allow-none annotations for g-i
2420 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2422 * gst/rtsp-server/rtsp-stream.c:
2423 * gst/rtsp-server/rtsp-stream.h:
2424 stream: add method to filter transports
2425 Add a method to safely iterate and collect the stream transports
2426 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2428 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2430 * gst/rtsp-server/rtsp-client.c:
2431 * gst/rtsp-server/rtsp-server.c:
2432 * gst/rtsp-server/rtsp-session-pool.c:
2433 * gst/rtsp-server/rtsp-session.c:
2434 rtsp: allow NULL func in filters
2435 Passing a null function make the filters return a list of
2438 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2440 * gst/rtsp-server/rtsp-address-pool.c:
2441 * tests/check/gst/addresspool.c:
2442 address-pool: fix address increment
2443 Use a guint instead of guint8 to increment the address. It's still not
2444 completely correct because a guint might not be able to hold the complete
2445 address range, but that's an enhacement for later.
2446 Add unit test to test improved behaviour.
2447 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2449 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2451 * gst/rtsp-server/rtsp-client.c:
2452 * tests/check/gst/client.c:
2453 client: allow absolute path in requests
2454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2456 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2458 * gst/rtsp-server/rtsp-client.c:
2459 * gst/rtsp-server/rtsp-client.h:
2460 client: make make_path_from_uri a vmethod
2462 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2464 * docs/libs/gst-rtsp-server-sections.txt:
2465 * gst/rtsp-server/rtsp-stream.c:
2466 * gst/rtsp-server/rtsp-stream.h:
2467 * tests/check/Makefile.am:
2468 * tests/check/gst/stream.c:
2469 stream: Add functions to get rtp and rtcp sockets
2470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2472 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2474 * gst/rtsp-server/rtsp-context.c:
2475 * gst/rtsp-server/rtsp-context.h:
2476 context: defing a GType for the context
2477 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2479 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2481 * gst/rtsp-server/Makefile.am:
2482 * gst/rtsp-server/rtsp-auth.c:
2483 * gst/rtsp-server/rtsp-context.c:
2484 * gst/rtsp-server/rtsp-media.c:
2485 * gst/rtsp-server/rtsp-mount-points.c:
2486 * gst/rtsp-server/rtsp-server.h:
2487 * gst/rtsp-server/rtsp-session-media.c:
2488 * gst/rtsp-server/rtsp-session.c:
2489 * gst/rtsp-server/rtsp-stream.c:
2490 Fixed several GIR warnings
2492 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2494 * gst/rtsp-server/rtsp-auth.c:
2497 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2499 * tests/check/Makefile.am:
2500 * tests/check/gst/token.c:
2501 tests: Add unit tests for token
2502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2504 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2506 * gst/rtsp-server/rtsp-token.c:
2507 token: Validate args for gst_rtsp_token_is_allowed
2508 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2510 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2512 * gst/rtsp-server/rtsp-token.c:
2513 token: Fix bug when creating empty token
2514 We always want to have a valid GstStructure in the token.
2515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2517 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2519 * gst/rtsp-server/rtsp-thread-pool.c:
2520 thread-pool: avoid race in shutdown
2521 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2522 don't actually stop the mainloop ever. Solve this race by adding an idle source
2523 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2524 if quit was called before we started it.
2526 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2528 * tests/check/Makefile.am:
2529 * tests/check/gst/permissions.c:
2530 tests: Add unit tests for permissions
2531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2533 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2535 * tests/check/gst/mediafactory.c:
2536 tests: Test mediafactory permissions
2537 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2539 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2541 * gst/rtsp-server/rtsp-permissions.c:
2542 permissions: Fix refcounting when adding/removing roles
2543 Previously a role that was removed was unreffed twice, and when
2544 replacing an existing role the replaced role was freed while still being
2545 referenced. Both bugs are now fixed.
2546 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2548 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2550 * tests/check/gst/media.c:
2551 * tests/check/gst/mediafactory.c:
2552 * tests/check/gst/rtspserver.c:
2553 tests: Check gst_rtsp_url_parse return value
2554 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2556 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2559 Automatic update of common submodule
2560 From 865aa20 to dbedaa0
2562 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2564 * gst/rtsp-server/rtsp-server.c:
2565 rtsp-server: Fix socket leak
2566 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2568 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2570 * gst/rtsp-server/rtsp-session-pool.c:
2571 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2572 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2574 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2576 * examples/.gitignore:
2577 * examples/test-video.c:
2578 examples: fix compilation when WITH_AUTH is defined
2579 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2581 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2584 gitignore: Add new test binary
2586 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2588 * tests/check/Makefile.am:
2589 * tests/check/gst/threadpool.c:
2590 thread-pool: Add unit test for the thread pools
2591 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2593 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2595 * gst/rtsp-server/rtsp-thread-pool.c:
2596 thread-pool: Fix thread leak when reusing threads
2597 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2599 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2601 * gst/rtsp-server/rtsp-server.c:
2602 * tests/check/gst/rtspserver.c:
2603 tests: fixed racy behavior in rtspserver tests
2604 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2606 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2608 * tests/check/gst/addresspool.c:
2609 tests: Improve address pool unit tests
2610 Add a range with mixed IPV4 and IPV6 addresses to pool.
2611 Get an IPV4 address from an IPV6-only pool.
2612 Get an IPV6 address from an IPV4-only pool.
2613 Reserve a IPV6 address from an IPV4-only pool.
2614 Check for unicast addresses in multicast-only pool.
2615 Check for unicast addresses in uni-/multicast-mixed pool.
2616 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2618 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2620 * gst/rtsp-server/rtsp-client.c:
2621 client: append query string in PAUSE/PLAY/TEARDOWN as well
2623 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2625 * gst/rtsp-server/rtsp-client.c:
2626 client: Add query to control path
2627 If the SETUP url contains a query it must be appended to the control
2628 path so that it matches any already created stream in the media. The
2629 query will also be appended to the session media path.
2631 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2633 * gst/rtsp-server/rtsp-media.c:
2634 rtsp-media: remove old line
2636 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2638 * gst/rtsp-server/rtsp-stream.c:
2639 stream: Correct control comparison
2640 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2642 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2644 * gst/rtsp-server/rtsp-media.c:
2645 media: Check dynamically if the pipeline supports seeking
2646 We should not depend on whether or not the pipeline state change
2647 returned NO_PREROLL or not. A media could dynamically change its
2648 element and switch from seekable to non seekable so it's best to test
2649 the seekable nature of the pipeline dynamically when we try to do a seek.
2651 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2653 * gst/rtsp-server/rtsp-media.c:
2654 media: Return FALSE if seeking is not supported
2656 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2658 * gst/rtsp-server/rtsp-media.c:
2659 rtsp-media: don't seek accurate by default
2660 Accurate seeking is perhaps a little overkill in the most common situation and
2661 causes some formats (mp3) over slow media to seek extremely slowly.
2663 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2665 * tests/check/gst/rtspserver.c:
2666 tests: fix unit test
2667 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2669 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2671 * gst/rtsp-server/rtsp-client.c:
2672 client: Reply 400 if media cannot be constructed
2673 Reply 400 Bad Request instead of 503 Service Unavailable if media
2674 cannot be constructed in SETUP.
2675 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2677 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2679 * gst/rtsp-server/rtsp-client.c:
2680 client: Send setup reply once only
2681 If find_media() failed in handle_setup_request() two replies was sent.
2682 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2684 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2687 Automatic update of common submodule
2688 From 6b03ba7 to 865aa20
2690 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2692 * gst/rtsp-server/rtsp-server.c:
2693 server: Emit client-connected signal earlier
2694 Emit client-connected before the client ref is given to a GSource,
2695 otherwise client-connected can be emitted after the client object has
2698 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2700 * gst/rtsp-server/rtsp-address-pool.c:
2701 * gst/rtsp-server/rtsp-address-pool.h:
2702 * gst/rtsp-server/rtsp-stream.c:
2703 * tests/check/gst/addresspool.c:
2704 addresspool: return reason of failure
2705 Let gst_rtsp_address_pool_reserve_address() return the reason why
2706 the address could not be reserved.
2707 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2709 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2712 autogen.sh: Sync behaviour with other GStreamer modules
2713 Allows building from outside of tree amongst other things
2715 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2718 Automatic update of common submodule
2719 From b613661 to 6b03ba7
2721 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2724 Automatic update of common submodule
2725 From 74a6857 to b613661
2727 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2730 Automatic update of common submodule
2731 From 01a7a46 to 74a6857
2733 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2735 * gst/rtsp-server/rtsp-client.c:
2736 client: Do not read beyond end of path string
2737 If the setup was done without a control url, make sure we don't try to read the
2738 non-existing control string and crash.
2740 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2742 * gst/rtsp-server/rtsp-client.c:
2743 client: Fix RTPInfo header
2744 Refactor the method to make the content_base.
2745 Use the content-base and the control url to construct the RTPInfo
2748 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2750 * gst/rtsp-server/rtsp-client.c:
2751 client: map url to path only in describe
2752 Only map the request url to a path in the DESCRIBE method. The SDP then
2753 contains the base and control urls that should be used to SETUP/PAUSE/
2754 PLAY/TEARDOWN the media.
2756 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2758 * gst/rtsp-server/rtsp-client.c:
2759 Revert "client: map URL to path in requests"
2760 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2761 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2762 contains the base and control urls which are used in the SETUP, PLAY,
2763 PAUSE and TEARDOWN requests.
2765 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2767 * gst/rtsp-server/rtsp-client.c:
2768 client: map URL to path in requests
2770 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2772 * gst/rtsp-server/rtsp-client.c:
2773 * gst/rtsp-server/rtsp-mount-points.c:
2774 * gst/rtsp-server/rtsp-mount-points.h:
2775 mount-points: make vmethod to make path from uri
2776 Make a vmethod to transform an url into a path. The path is then used to lookup
2777 the factory. This makes it possible to also use other bits of the url, such as
2778 the query parameters, to locate the factory.
2780 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2782 * gst/rtsp-server/rtsp-thread-pool.c:
2783 * gst/rtsp-server/rtsp-thread-pool.h:
2784 thread-pool: Add cleanup to wait for the threadpool to finish
2785 Also fix race condition if two threads are asking for the first
2786 thread from the thread pool at once. This would case two internal
2787 GThreadPools to be created.
2788 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2790 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2792 * gst/rtsp-server/rtsp-client.c:
2793 * tests/check/gst/client.c:
2794 client: free threadpool
2795 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2797 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2799 * tests/check/gst/mountpoints.c:
2800 mountpoints tests: unref matched factories
2801 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2803 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2805 * tests/check/gst/media.c:
2806 media tests: unref thread pool and caps
2807 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2809 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2811 * gst/rtsp-server/rtsp-auth.c:
2812 * gst/rtsp-server/rtsp-media-factory.c:
2813 * gst/rtsp-server/rtsp-media.c:
2814 auth, media, media-factory: unref permissions
2815 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2817 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2819 * examples/Makefile.am:
2820 Makefile: add rule for appsrc example
2822 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2824 * examples/test-appsrc.c:
2825 tests: add appsrc example
2826 Add an example on how to use appsrc to feed the server pipeline with data.
2828 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2830 * gst/rtsp-server/rtsp-client.c:
2831 rtsp-client: remove query part from content-base string
2832 Make sure that after the control url has been resolved, it's
2833 not a part of the query-string.
2834 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2836 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2838 * gst/rtsp-server/rtsp-client.c:
2839 client: don't check url in response
2840 There is no url or method in the response to check
2842 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2844 * gst/rtsp-server/rtsp-client.c:
2845 * gst/rtsp-server/rtsp-client.h:
2846 Add handle-response signal for when we receive a GET_PARAMETER response
2848 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2850 * gst/rtsp-server/rtsp-server.c:
2851 Fix gst_rtsp_server_client_filter, using wrong variable type
2853 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2855 * gst/rtsp-server/rtsp-media-factory-uri.c:
2856 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2857 For AAC we need to check for framed=true instead of parsed=true.
2858 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2860 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2862 * gst/rtsp-server/rtsp-stream.c:
2863 stream: optimize pipeline for protocols
2864 When TCP is not an allowed protocol for the stream, avoid creating the
2865 appsrc/appsink/queue and tee elements.
2867 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2869 * gst/rtsp-server/rtsp-media.c:
2870 media: set protocols on streams
2872 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2874 * gst/rtsp-server/rtsp-client.c:
2875 client: use protocols supported by stream
2877 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2879 * gst/rtsp-server/rtsp-media-factory.c:
2880 * gst/rtsp-server/rtsp-media.c:
2881 * gst/rtsp-server/rtsp-stream.c:
2882 media-factory: allow all protocols
2884 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2886 * gst/rtsp-server/rtsp-media.c:
2887 media: configure protocols in new streams
2889 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2891 * gst/rtsp-server/rtsp-stream.c:
2892 * gst/rtsp-server/rtsp-stream.h:
2893 stream: add protocols property
2895 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2897 * gst/rtsp-server/rtsp-media.c:
2898 rtsp-media: send state in "new-state" signal
2899 https://bugzilla.gnome.org/show_bug.cgi?id=705110
2901 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
2904 build: add subdir-objects to AM_INIT_AUTOMAKE
2905 Fixes warnings with automake 1.14
2906 https://bugzilla.gnome.org/show_bug.cgi?id=705350
2908 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2910 * docs/libs/gst-rtsp-server-sections.txt:
2911 * gst/rtsp-server/rtsp-client.c:
2912 * gst/rtsp-server/rtsp-server.c:
2913 * gst/rtsp-server/rtsp-server.h:
2914 server: add method to iterate clients of server
2916 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2918 * gst/rtsp-server/rtsp-media.c:
2919 * gst/rtsp-server/rtsp-media.h:
2920 Add vmethod for rtsp-media subclass to access rtpbin
2922 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2924 * gst/rtsp-server/rtsp-client.h:
2925 small documentation fix
2927 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2929 * gst/rtsp-server/rtsp-client.c:
2930 Do not take range header if range is invalid
2932 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2934 * docs/libs/gst-rtsp-server-sections.txt:
2935 * gst/rtsp-server/rtsp-media.c:
2936 media: add docs for new method
2938 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2940 * gst/rtsp-server/rtsp-media.c:
2941 * gst/rtsp-server/rtsp-media.h:
2942 Add API to rtsp-media set the pipeline's state
2944 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2946 * gst/rtsp-server/rtsp-media.c:
2947 Update current position/duration when gst_rtsp_media_get_range_string is called
2949 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2951 * examples/test-cgroups.c:
2952 tests: add some more docs
2954 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2956 * examples/test-cgroups.c:
2957 * gst/rtsp-server/Makefile.am:
2958 * gst/rtsp-server/rtsp-auth.c:
2959 * gst/rtsp-server/rtsp-auth.h:
2960 * gst/rtsp-server/rtsp-client.c:
2961 * gst/rtsp-server/rtsp-client.h:
2962 * gst/rtsp-server/rtsp-context.c:
2963 * gst/rtsp-server/rtsp-context.h:
2964 * gst/rtsp-server/rtsp-params.c:
2965 * gst/rtsp-server/rtsp-params.h:
2966 * gst/rtsp-server/rtsp-server.c:
2967 * gst/rtsp-server/rtsp-thread-pool.c:
2968 * gst/rtsp-server/rtsp-thread-pool.h:
2969 * tests/check/gst/client.c:
2970 ClientState -> Context
2971 Rename the clientstate to context and put the code in a separate file.
2973 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2975 * examples/test-auth.c:
2976 * gst/rtsp-server/rtsp-auth.c:
2977 * gst/rtsp-server/rtsp-auth.h:
2978 auth: add support for default token
2979 The default token is used when the user is not authenticated and can be used to
2980 give minimal permissions.
2982 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2984 * examples/test-auth.c:
2985 * gst/rtsp-server/rtsp-auth.c:
2986 auth: use defines when possible
2988 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2990 * gst/rtsp-server/rtsp-address-pool.c:
2991 address-pool: improve docs
2993 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2995 * gst/rtsp-server/rtsp-permissions.c:
2996 permissions: add the role to the copy
2998 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3000 * gst/rtsp-server/rtsp-permissions.c:
3001 permissions: Also copy the roles
3003 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3005 * gst/rtsp-server/rtsp-permissions.c:
3006 permissions: Make it build
3008 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3010 * gst/rtsp-server/rtsp-address-pool.h:
3013 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3015 * docs/libs/gst-rtsp-server-sections.txt:
3016 * gst/rtsp-server/rtsp-auth.c:
3017 * gst/rtsp-server/rtsp-auth.h:
3018 * gst/rtsp-server/rtsp-media.c:
3019 * gst/rtsp-server/rtsp-session-media.c:
3020 * gst/rtsp-server/rtsp-stream-transport.c:
3021 * gst/rtsp-server/rtsp-stream-transport.h:
3022 * gst/rtsp-server/rtsp-stream.c:
3023 * tests/check/gst/client.c:
3026 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3028 * docs/libs/gst-rtsp-server-sections.txt:
3029 * gst/rtsp-server/rtsp-address-pool.c:
3030 * gst/rtsp-server/rtsp-address-pool.h:
3031 * tests/check/gst/addresspool.c:
3032 * tests/check/gst/rtspserver.c:
3033 address-pool: cleanups
3034 Remove redundant method, improve docs.
3036 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3038 * docs/libs/gst-rtsp-server-sections.txt:
3039 * gst/rtsp-server/rtsp-auth.h:
3040 * gst/rtsp-server/rtsp-permissions.c:
3041 * gst/rtsp-server/rtsp-permissions.h:
3042 * gst/rtsp-server/rtsp-token.c:
3045 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3047 * gst/rtsp-server/rtsp-permissions.c:
3048 permissions: implement _remove_role
3050 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3052 * gst/rtsp-server/rtsp-permissions.c:
3053 permissions: update docs
3055 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3057 * tests/check/gst/client.c:
3058 tests: simplify tests
3059 Client settings are now disabled by default so we don't need an auth
3060 module to disable them.
3062 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3064 * gst/rtsp-server/rtsp-auth.c:
3065 auth: add default authorizations
3066 When no auth module is specified, use our table of defaults to look up the
3067 default value of the check instead of always allowing everything. This was
3068 we can disallow client settings by default.
3070 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3073 README: update readme
3075 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3077 * gst/rtsp-server/rtsp-thread-pool.c:
3078 * gst/rtsp-server/rtsp-thread-pool.h:
3079 thread-pool: add more docs
3081 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3083 * gst/rtsp-server/rtsp-thread-pool.c:
3084 * gst/rtsp-server/rtsp-thread-pool.h:
3085 thread-pool: fix race in thread reuse
3086 If we try to reuse a thread right after we made it stop, we end up using a
3087 stopped thread. Catch this case and only reuse threads that are not stopping.
3089 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3091 * gst/rtsp-server/rtsp-server.c:
3092 server: add small debug
3094 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3096 * tests/check/gst/client.c:
3098 Add some permissions to media so we can use the auth and enable
3101 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3103 * gst/rtsp-server/rtsp-client.c:
3104 client: support pushed context in handle_request
3105 If we already have a pushed state, reuse it and add our own things. This makes
3106 it easier to write tests.
3108 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3110 * gst/rtsp-server/rtsp-auth.c:
3111 auth: don't auth on methods
3112 Don't authorize on methods anymore but on the resources that we
3113 try to access, this is more flexible.
3114 Move the authorization checks to where they are needed and let the
3115 check return the response on error.
3117 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3119 * gst/rtsp-server/rtsp-mount-points.c:
3120 mount-points: add some debug
3122 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3124 * tests/check/gst/client.c:
3125 tests: almost fix test
3127 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3129 * gst/rtsp-server/rtsp-auth.c:
3130 * gst/rtsp-server/rtsp-auth.h:
3131 * gst/rtsp-server/rtsp-client.c:
3132 * gst/rtsp-server/rtsp-client.h:
3133 * gst/rtsp-server/rtsp-server.c:
3134 * gst/rtsp-server/rtsp-server.h:
3135 auth: let the auth module check client_settings
3136 Let the auth module decide if client settings are allowed for the
3139 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3141 * gst/rtsp-server/rtsp-token.c:
3142 * gst/rtsp-server/rtsp-token.h:
3143 token: add method to check boolean permission
3145 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3147 * examples/test-auth.c:
3148 * examples/test-cgroups.c:
3149 * gst/rtsp-server/rtsp-token.c:
3150 * gst/rtsp-server/rtsp-token.h:
3151 token: simplify token constructor
3152 Use variable arguments to make easier API.
3154 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3156 * examples/test-auth.c:
3157 * examples/test-cgroups.c:
3158 * gst/rtsp-server/rtsp-media-factory.c:
3159 * gst/rtsp-server/rtsp-media-factory.h:
3160 media-factory: add convenience API for factory
3162 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3164 * examples/test-auth.c:
3165 * examples/test-cgroups.c:
3166 * gst/rtsp-server/rtsp-permissions.c:
3167 * gst/rtsp-server/rtsp-permissions.h:
3168 permissions: simplify API a little
3169 Avoid passing GstStructure in the add_role method, use varargs instead
3170 to construct the structure behind the scenes. We can then also use the
3171 structure name as the role and simplify some more logic.
3173 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3175 * gst/rtsp-server/rtsp-auth.c:
3178 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3180 * gst/rtsp-server/rtsp-auth.c:
3181 * gst/rtsp-server/rtsp-auth.h:
3182 * gst/rtsp-server/rtsp-client.c:
3183 auth: handle unauthorized response
3184 Move handling of the unauthorized response to the auth module, it can add
3185 the appropriate headers to request authorization for the required method
3186 much better than the client.
3188 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3190 * gst/rtsp-server/rtsp-client.c:
3191 * gst/rtsp-server/rtsp-client.h:
3192 client: allow for sending any message, not only requests
3193 Change the _send_request() method to _send_message() so that we
3194 can both send requests and replies.
3196 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3198 * docs/libs/gst-rtsp-server-sections.txt:
3199 * gst/rtsp-server/rtsp-server.h:
3202 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3204 * examples/test-video.c:
3205 * gst/rtsp-server/rtsp-auth.c:
3206 * gst/rtsp-server/rtsp-auth.h:
3207 * gst/rtsp-server/rtsp-server.c:
3208 * gst/rtsp-server/rtsp-server.h:
3209 auth: move TLS handling to auth module
3210 Remove the TLS settings on the server and move it to the auth module because
3211 that is where security related bits go.
3213 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3215 * gst/rtsp-server/rtsp-client.c:
3216 * gst/rtsp-server/rtsp-client.h:
3217 client: add state push/pop
3219 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3221 * gst/rtsp-server/rtsp-client.c:
3222 * gst/rtsp-server/rtsp-client.h:
3223 client: add connection to state
3225 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3227 * gst/rtsp-server/rtsp-mount-points.c:
3228 mount-points: fix debug
3230 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3232 * tests/check/gst/media.c:
3233 tests: fix media test
3235 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3237 * gst/rtsp-server/rtsp-thread-pool.c:
3238 thread-pool: we don't require a state
3240 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3242 * gst/rtsp-server/rtsp-server.c:
3243 server: let context ref the server
3244 So that we don't risk losing the server object early anc crash.
3246 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3248 * tests/check/gst/client.c:
3249 tests: fix client test
3251 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3254 * docs/libs/gst-rtsp-server-docs.sgml:
3255 * docs/libs/gst-rtsp-server-sections.txt:
3256 * gst/rtsp-server/rtsp-address-pool.c:
3257 * gst/rtsp-server/rtsp-auth.c:
3258 * gst/rtsp-server/rtsp-client.c:
3259 * gst/rtsp-server/rtsp-client.h:
3260 * gst/rtsp-server/rtsp-media-factory-uri.c:
3261 * gst/rtsp-server/rtsp-media-factory.c:
3262 * gst/rtsp-server/rtsp-media-factory.h:
3263 * gst/rtsp-server/rtsp-media.c:
3264 * gst/rtsp-server/rtsp-mount-points.c:
3265 * gst/rtsp-server/rtsp-params.c:
3266 * gst/rtsp-server/rtsp-permissions.c:
3267 * gst/rtsp-server/rtsp-sdp.c:
3268 * gst/rtsp-server/rtsp-server.c:
3269 * gst/rtsp-server/rtsp-server.h:
3270 * gst/rtsp-server/rtsp-session-media.c:
3271 * gst/rtsp-server/rtsp-session-pool.c:
3272 * gst/rtsp-server/rtsp-session.c:
3273 * gst/rtsp-server/rtsp-stream-transport.c:
3274 * gst/rtsp-server/rtsp-stream.c:
3275 * gst/rtsp-server/rtsp-thread-pool.c:
3276 * gst/rtsp-server/rtsp-token.c:
3279 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3281 * gst/rtsp-server/rtsp-session-pool.c:
3282 * gst/rtsp-server/rtsp-session-pool.h:
3283 session-pool: make vmethod to create a session
3284 Make a vmethod to create a sessions so that subclasses can create
3285 custom session objects
3287 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3289 * gst/rtsp-server/rtsp-auth.c:
3290 * gst/rtsp-server/rtsp-media-factory.h:
3291 * gst/rtsp-server/rtsp-media.h:
3292 * gst/rtsp-server/rtsp-mount-points.h:
3293 * gst/rtsp-server/rtsp-session-pool.h:
3294 * gst/rtsp-server/rtsp-stream.h:
3297 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3299 * docs/libs/gst-rtsp-server-docs.sgml:
3300 * docs/libs/gst-rtsp-server-sections.txt:
3301 * gst/rtsp-server/rtsp-address-pool.c:
3302 * gst/rtsp-server/rtsp-address-pool.h:
3303 * gst/rtsp-server/rtsp-auth.c:
3304 * gst/rtsp-server/rtsp-client.h:
3305 * gst/rtsp-server/rtsp-media-factory.h:
3306 * gst/rtsp-server/rtsp-media.c:
3307 * gst/rtsp-server/rtsp-media.h:
3308 * gst/rtsp-server/rtsp-permissions.c:
3309 * gst/rtsp-server/rtsp-permissions.h:
3310 * gst/rtsp-server/rtsp-server.h:
3311 * gst/rtsp-server/rtsp-session-media.c:
3312 * gst/rtsp-server/rtsp-session-media.h:
3313 * gst/rtsp-server/rtsp-session-pool.h:
3314 * gst/rtsp-server/rtsp-session.h:
3315 * gst/rtsp-server/rtsp-stream-transport.h:
3316 * gst/rtsp-server/rtsp-stream.c:
3317 * gst/rtsp-server/rtsp-thread-pool.h:
3320 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3323 * examples/Makefile.am:
3324 configure: compile cgroup example conditionally
3325 Only compile the cgroup example when we have libcgroup
3327 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3330 * examples/Makefile.am:
3331 * examples/test-cgroups.c:
3332 examples: add cgroups example
3334 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3336 * tests/check/gst/rtspserver.c:
3337 tests: fix compilation
3339 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3341 * gst/rtsp-server/rtsp-thread-pool.c:
3342 thread-pool: fix vmethod invocation
3344 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3346 * gst/rtsp-server/rtsp-thread-pool.c:
3347 * gst/rtsp-server/rtsp-thread-pool.h:
3348 thread-pool: store thread type in thread
3350 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3352 * gst/rtsp-server/rtsp-client.c:
3353 client: pass thread from pool to media _prepare
3354 Get a thread from the configured threadpool and pass it to the prepare method of
3357 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3359 * gst/rtsp-server/rtsp-media.c:
3360 * gst/rtsp-server/rtsp-media.h:
3361 media: Accept a thread in _prepare
3362 Remove out own threadpool handling and use the provided thread and
3363 maincontext for the bus messages and the state changes.
3365 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3367 * gst/rtsp-server/rtsp-server.c:
3368 server: configure client thread pool
3370 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3372 * gst/rtsp-server/rtsp-client.c:
3373 * gst/rtsp-server/rtsp-client.h:
3374 client: add method to configure thread pool
3376 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3378 * gst/rtsp-server/rtsp-client.h:
3379 * gst/rtsp-server/rtsp-server.c:
3380 * gst/rtsp-server/rtsp-server.h:
3381 server: use thread pool
3382 Use the thread pool instead of doing our own thing.
3384 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3386 * gst/rtsp-server/Makefile.am:
3387 * gst/rtsp-server/rtsp-thread-pool.c:
3388 * gst/rtsp-server/rtsp-thread-pool.h:
3389 thread-pool: add object to manage threads
3390 Add an object to manage the client and media threads.
3392 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3394 * gst/rtsp-server/rtsp-auth.c:
3395 auth: debug authorization check
3397 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3399 * gst/rtsp-server/rtsp-media.c:
3400 media: start media pipeline in context
3401 Start the media pipeline in the provided context (or our default one
3402 when NULL). This makes sure that we run the bus thread in this context and that
3403 all media threads are children of this context.
3405 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3407 * gst/rtsp-server/rtsp-media-factory.c:
3408 factory: pass permissions to media by default
3410 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3412 * examples/test-auth.c:
3413 test: add permissions to auth test
3414 Ass some permissions to the media factory in the test.
3416 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3418 * gst/rtsp-server/rtsp-auth.c:
3419 * gst/rtsp-server/rtsp-auth.h:
3420 * gst/rtsp-server/rtsp-client.c:
3421 auth: simplify auth checks
3422 Remove client from methods, it's now in the state
3423 Perform the check specified by the string, use the information from the
3424 thread local context.
3426 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3428 * gst/rtsp-server/rtsp-client.c:
3429 * gst/rtsp-server/rtsp-client.h:
3430 client: add state to current thread
3431 Add the client to the ClientState object.
3432 Place the ClientState on the current thread.
3434 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3436 * gst/rtsp-server/rtsp-media-factory.c:
3437 * gst/rtsp-server/rtsp-media-factory.h:
3438 * gst/rtsp-server/rtsp-media.c:
3439 * gst/rtsp-server/rtsp-media.h:
3440 media: make it possible to set permissions
3441 Make it possible to set permissions on media and media factory objects
3443 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3445 * gst/rtsp-server/Makefile.am:
3446 * gst/rtsp-server/rtsp-permissions.c:
3447 * gst/rtsp-server/rtsp-permissions.h:
3448 permissions: add permissions object
3449 Add a mini object to store permissions based on a role.
3451 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3453 * examples/test-auth.c:
3454 * gst/rtsp-server/rtsp-auth.c:
3455 * gst/rtsp-server/rtsp-auth.h:
3456 * gst/rtsp-server/rtsp-client.c:
3457 auth: add auth checks
3458 Add an enum with auth checks and implement the checks in the auth object.
3459 Perform the checks from the client.
3461 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3463 * examples/test-auth.c:
3464 * gst/rtsp-server/rtsp-auth.c:
3465 * gst/rtsp-server/rtsp-auth.h:
3466 * gst/rtsp-server/rtsp-client.h:
3467 auth: use the token after authentication
3468 After we authenticated a user, keep the Token around in the state.
3470 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3472 * gst/rtsp-server/rtsp-client.c:
3473 * gst/rtsp-server/rtsp-media.c:
3474 * gst/rtsp-server/rtsp-media.h:
3475 * tests/check/gst/media.c:
3476 media: add optional context for bus messages
3477 Add an optional mainloop to _prepare that will handle the bus messages instead
3478 of always using the shared mainloop.
3480 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/rtsp-server/Makefile.am:
3483 * gst/rtsp-server/rtsp-token.c:
3484 * gst/rtsp-server/rtsp-token.h:
3485 token: add authorization token
3486 Add a simply miniobject that contains the authorizations. The object contains a
3487 GstStructure that hold all authorization fields. When a user is authenticated,
3488 the auth module will create a Token for the user. The token is then used to
3489 check what operations the user is allowed to do and various other configuration
3492 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3494 * examples/test-auth.c:
3495 * gst/rtsp-server/rtsp-auth.c:
3496 * gst/rtsp-server/rtsp-auth.h:
3497 * gst/rtsp-server/rtsp-client.c:
3498 * gst/rtsp-server/rtsp-client.h:
3499 * gst/rtsp-server/rtsp-media-factory.c:
3500 * gst/rtsp-server/rtsp-media-factory.h:
3501 * gst/rtsp-server/rtsp-media.c:
3502 * gst/rtsp-server/rtsp-media.h:
3503 auth: remove auth from media and factory
3504 Remove the auth object from media and factory. We want to have the RTSPClient
3505 authenticate and authorize resources, there is no need to place another auth
3506 manager on the media/factory.
3508 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3510 * examples/test-auth.c:
3511 * gst/rtsp-server/rtsp-auth.c:
3512 * gst/rtsp-server/rtsp-auth.h:
3513 * gst/rtsp-server/rtsp-client.h:
3514 auth: add support for multiple basic auth tokens
3515 Make it possible to add multiple basic authorisation tokens to one authorization
3516 object. Associate with each token an authorization group that will define what
3517 capabilities are allowed.
3519 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * gst/rtsp-server/rtsp-client.c:
3522 client: error out on non-aggregate control
3523 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3525 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3527 * gst/rtsp-server/rtsp-client.c:
3528 client: rework setup request a little
3529 Cache the media in DESCRIBE based on the longest matching path with the uri
3530 that we can find in the mount points.
3531 Rework the setup request a little to get the media from the session or from
3532 the longest matching path, this way we can derive the control string as
3533 everything after the path instead of hardcoding it.
3534 Find the stream based on the control string and only open a session when all
3537 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3539 * gst/rtsp-server/rtsp-media.c:
3540 * gst/rtsp-server/rtsp-media.h:
3541 media: add method to find a stream by control url
3543 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3545 * gst/rtsp-server/rtsp-stream.c:
3546 * gst/rtsp-server/rtsp-stream.h:
3547 stream: add method to check control url of stream
3549 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3551 * gst/rtsp-server/rtsp-client.c:
3552 * gst/rtsp-server/rtsp-session-media.c:
3553 * gst/rtsp-server/rtsp-session-media.h:
3554 * gst/rtsp-server/rtsp-session.c:
3555 * gst/rtsp-server/rtsp-session.h:
3556 session: use path matching for session media
3557 Use a path string instead of a uri to lookup session media in the sessions. Also
3558 use path matching to find the largest possible path that matches.
3560 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3562 * gst/rtsp-server/rtsp-client.c:
3563 * gst/rtsp-server/rtsp-mount-points.c:
3564 * gst/rtsp-server/rtsp-mount-points.h:
3565 * tests/check/gst/mountpoints.c:
3566 mount-points: remove useless vmethod
3567 Making lookups in the mount points should not be done with a URL, if there is a
3568 mapping to be done from URL to mount points, we'll need to do it somewhere
3571 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * gst/rtsp-server/rtsp-mount-points.c:
3574 * gst/rtsp-server/rtsp-mount-points.h:
3575 * tests/check/gst/mountpoints.c:
3576 mount-points: improve mount point searching
3577 Use a GSequence to keep track of the mount points.
3578 Match a URL to the longest matching registered mount point. This should be the
3579 URL to perform aggreagate control and the remainder is the stream specific
3581 Add some unit tests for this.
3583 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3585 * gst/rtsp-server/Makefile.am:
3586 rtsp-server: Allow building of static library
3588 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3590 * tests/check/gst/mediafactory.c:
3591 tests: fix compilation
3593 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3595 * gst/rtsp-server/rtsp-sdp.c:
3596 sdp: get control string from stream
3597 Use the control string as configured in the stream.
3599 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3601 * gst/rtsp-server/rtsp-stream.c:
3602 * gst/rtsp-server/rtsp-stream.h:
3603 stream: add methods and property to set control string
3605 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3607 * gst/rtsp-server/rtsp-client.c:
3609 Rename variables for clarity
3610 Keep media in state when we can
3612 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3614 * gst/rtsp-server/rtsp-client.c:
3615 * gst/rtsp-server/rtsp-stream.c:
3616 * gst/rtsp-server/rtsp-stream.h:
3617 stream: add more support for IPv6
3618 Rename _get_address to _get_multicast_address in GstRTSPStream to
3619 make it clear that this function only deals with multicast.
3620 Make it possible to have both an IPv4 and IPv6 multicast address on
3621 a stream. Give the client an IPv4 or IPv6 address depending on the
3622 address it used to connect to the server.
3623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3625 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3627 * gst/rtsp-server/rtsp-client.c:
3630 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3632 * gst/rtsp-server/rtsp-stream.c:
3633 stream: handle failed port allocation
3634 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3635 can't allocate any family at all. Also keep track of what port families we
3637 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3639 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3641 * gst/rtsp-server/rtsp-stream.c:
3642 stream: improve docs
3644 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3646 * gst/rtsp-server/rtsp-stream-transport.c:
3647 stream-transport: remove old if 0 block
3649 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3651 * tests/check/gst/client.c:
3653 gst_rtsp_client_get_uri() has been removed
3654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3656 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3658 * gst/rtsp-server/rtsp-client.c:
3659 * gst/rtsp-server/rtsp-client.h:
3660 client: add method to filter managed sessions
3661 Add a method to filter the sessions managed by this client connection.
3662 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3664 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3666 * gst/rtsp-server/rtsp-client.c:
3667 * gst/rtsp-server/rtsp-client.h:
3668 client: remove _get_uri() method
3669 Remove the get_uri() method on the client. A client has no uri, the uri
3670 property is an internal property to manage the last cached media for
3673 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3675 * gst/rtsp-server/rtsp-media-factory.h:
3676 media-factory: fix typo
3678 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3680 * gst/rtsp-server/rtsp-media.c:
3681 rtsp-media: Do not leak the query in default_query_stop
3682 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3684 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3686 * gst/rtsp-server/rtsp-media.c:
3687 media: don't unlock when conversion fails
3688 Don't unlock the state lock when conversion fails because it was not locked.
3690 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3692 * gst/rtsp-server/rtsp-media.c:
3693 * gst/rtsp-server/rtsp-media.h:
3694 Add query_position and query_stop vmethods to rtsp-media
3696 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3698 * gst/rtsp-server/rtsp-media.c:
3699 Fix typo in property install for rtsp-media's time-provider
3701 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3703 * gst/rtsp-server/rtsp-client.c:
3704 * gst/rtsp-server/rtsp-client.h:
3705 client: clean some variables
3706 Clean some variables and add some guards to _send_request()
3708 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3710 * gst/rtsp-server/rtsp-client.c:
3711 * gst/rtsp-server/rtsp-client.h:
3712 Add gst_rtsp_client_send_request API
3713 This makes it possible to send arbitrary messages to a client, such as
3714 SET_PARAMETER or GET_PARAMETER
3716 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3718 * gst/rtsp-server/rtsp-media.c:
3719 * gst/rtsp-server/rtsp-media.h:
3720 media: add _get_element() method
3721 Add method to get the element used when creating the media.
3722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3724 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3726 * gst/rtsp-server/rtsp-media.c:
3729 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3731 * gst/rtsp-server/rtsp-stream.c:
3732 * gst/rtsp-server/rtsp-stream.h:
3733 stream: allow access to the rtp session
3734 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3736 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3738 * gst/rtsp-server/rtsp-stream.c:
3739 * gst/rtsp-server/rtsp-stream.h:
3740 dscp qos support in gst-rtsp-stream
3741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3743 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3745 * tests/check/gst/rtspserver.c:
3747 Actually do what the comment says. Also keep the old code around, not sure what
3748 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3749 it currently doesn't.
3751 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3753 * gst/rtsp-server/rtsp-client.c:
3754 client: also watch newly created session
3755 When we newly created a session, start watching it immediately instead of
3756 on the next request.
3758 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3760 * tests/check/gst/client.c:
3761 tests: add unit test for new-session
3762 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3764 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3766 * gst/rtsp-server/rtsp-client.c:
3767 client: emit new-session when new session is created
3768 Only emit new-session when we created a new session for a client, not when a
3769 client picked up a previous session.
3770 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3772 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3774 * gst/rtsp-server/rtsp-client.c:
3775 client: handle asterisk as path in requests
3776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3778 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * gst/rtsp-server/rtsp-media.c:
3781 media: handle segment query format mismatch
3782 It's possible that the segment query returns with a different format than what
3783 we asked for, handle this case also.
3785 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3787 * gst/rtsp-server/rtsp-media.c:
3788 media: use segment stop in collect_media_stats
3789 Use segment stop instead of duration as range end point.
3790 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3792 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3794 * gst/rtsp-server/rtsp-media.c:
3795 * tests/check/gst/media.c:
3796 rtsp-media: Do not leak the element in take_pipeline
3797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3799 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3801 * gst/rtsp-server/rtsp-client.c:
3802 * gst/rtsp-server/rtsp-client.h:
3803 rtsp-client: Make configure_client_transport virtual
3804 This patch makes configure_client_transport virtual. The functionality is
3805 needed to handle some weird clients sending multicast transport settings as url
3807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3809 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3811 * gst/rtsp-server/rtsp-client.c:
3812 * gst/rtsp-server/rtsp-client.h:
3813 rtsp-client: Make param_set and param_get virtual
3814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3816 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3818 * gst/rtsp-server/rtsp-client.c:
3819 * gst/rtsp-server/rtsp-media.c:
3820 * gst/rtsp-server/rtsp-media.h:
3821 media: convert_range replaces get_range_times
3822 get_range_times worked for handling UTC ranges for seeks, but we also
3823 need to convert back from NPT to the requested unit in
3824 get_range_string. convert_range is now used for both.
3825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3827 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3829 * gst/rtsp-server/rtsp-client.c:
3830 * gst/rtsp-server/rtsp-sdp.c:
3831 * gst/rtsp-server/rtsp-sdp.h:
3832 sdp: cleanup sdp info
3833 We don't need to pass the proto, we can more easily check a boolean.
3834 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3836 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3838 * gst/rtsp-server/rtsp-sdp.c:
3839 use 0.0.0.0 or :: for c= line instead of server address
3841 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3843 * gst/rtsp-server/rtsp-client.c:
3844 use local address, not remote, in SDP
3845 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3847 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3850 Automatic update of common submodule
3851 From 098c0d7 to 01a7a46
3853 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3855 * gst/rtsp-server/rtsp-media.c:
3856 * gst/rtsp-server/rtsp-media.h:
3857 media: possibility to override range time conversion
3858 Make it possible to override the conversion from GstRTSPTimeRange to
3859 GstClockTimes, that is done before seeking on the media
3860 pipeline. Overriding can be useful for UTC ranges, where the default
3861 conversion gives nanoseconds since 1900.
3862 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3864 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3866 * gst/rtsp-server/rtsp-server.c:
3867 * gst/rtsp-server/rtsp-server.h:
3868 rtsp-server: Expose the use_client_settings API
3869 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3871 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3873 * gst/rtsp-server/rtsp-client.c:
3874 * gst/rtsp-server/rtsp-stream.c:
3875 * gst/rtsp-server/rtsp-stream.h:
3876 rtspstream: handle both ipv4 and ipv6 clients
3877 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3879 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3881 * gst/rtsp-server/rtsp-sdp.c:
3882 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3883 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3884 We already have a way to place extra attributes in the SDP by using a string
3885 property with prefix x- or a- in the caps.
3887 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3889 * gst/rtsp-server/rtsp-sdp.c:
3890 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3891 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3892 We already have a way to place extra attributes in the SDP, just make a string
3893 property in the payloader with a- or x- prefix.
3895 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3897 * gst/rtsp-server/rtsp-sdp.c:
3898 rtsp: place a- and x- properties as attributes
3899 application/x-rtp has properties with a- and x- prefixes that should be
3900 placed as attributes in the SDP for the media instead of being added to the
3903 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3905 * examples/Makefile.am:
3906 * examples/test-video.c:
3907 example: add TLS example
3909 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3911 * gst/rtsp-server/rtsp-server.c:
3912 * gst/rtsp-server/rtsp-server.h:
3913 server: add support for TLS
3914 Add methods to set and get a TLS certificate.
3915 Add vmethod to configure a new connection. By default, configure the TLS
3916 certificate in a new connection if needed.
3918 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3920 * gst/rtsp-server/rtsp-server.c:
3921 * gst/rtsp-server/rtsp-server.h:
3922 server: remove accept_client vmethod
3923 This vmethod is not very useful so remove it.
3925 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3927 * gst/rtsp-server/rtsp-server.c:
3928 server: don't crash on NULL GError
3930 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
3932 * gst/rtsp-server/rtsp-session-pool.c:
3933 rtsp-session-pool: corrected session timeout detection
3934 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
3936 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3938 * gst/rtsp-server/rtsp-client.c:
3939 client: improve debug
3941 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3943 * gst/rtsp-server/rtsp-client.c:
3944 * gst/rtsp-server/rtsp-client.h:
3945 * gst/rtsp-server/rtsp-server.c:
3946 server: refactor connection setup
3947 Let the server accept the socket connection and construct a GstRTSPConnection
3948 from it. Remove the code from the client and let the client only deal with
3949 a fully configure GstRTSPConnection object.
3950 We will need this later when the server will configure the connection for
3953 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3955 * gst/rtsp-server/rtsp-stream.c:
3956 stream: keep the transport object alive
3957 Keep the transport object alive while we have it as qdata on the
3960 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
3962 * gst/rtsp-server/rtsp-client.c:
3963 * gst/rtsp-server/rtsp-server.c:
3964 rtsp-server: Do not crash on nmapping of server
3965 * generate error when gst_rtsp_connection_accept fails
3966 * do not stop accepting incoming connections because
3967 accepting a client fails
3968 https://bugzilla.gnome.org/show_bug.cgi?id=701072
3970 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
3972 * gst/rtsp-server/rtsp-client.c:
3973 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
3974 https://bugzilla.gnome.org/show_bug.cgi?id=700953
3976 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
3978 * gst/rtsp-server/rtsp-sdp.c:
3979 rtsp-sdp: Parse framerate caps field and set SDP attribute
3980 The SDP attribute and its format is described in RFC4566.
3981 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3983 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
3985 * gst/rtsp-server/rtsp-sdp.c:
3986 rtsp-sdp: Parse width/height from caps and set SDP attribute
3987 The SDP attribute and its format is described in RFC6064.
3988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3990 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
3992 * gst/rtsp-server/rtsp-sdp.c:
3993 * tests/check/gst/client.c:
3994 rtsp-sdp: add bandwidth line
3995 https://bugzilla.gnome.org/show_bug.cgi?id=699220
3997 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4000 Automatic update of common submodule
4001 From 5edcd85 to 098c0d7
4003 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4005 * tests/check/gst/media.c:
4006 tests: add dynamic payloader prepare/unprepare check
4008 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4010 * gst/rtsp-server/rtsp-media.c:
4011 media: release lock when removing fakesink
4013 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4015 * gst/rtsp-server/rtsp-stream.c:
4016 stream: set elements to NULL before removing
4017 When removing a stream, set the elements to NULL first. This avoids
4018 element-is-not-in-NULL-state errors when we dispose the elements.
4020 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4023 Automatic update of common submodule
4024 From 3cb3d3c to 5edcd85
4026 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4028 * gst/rtsp-server/rtsp-media.c:
4029 * gst/rtsp-server/rtsp-media.h:
4030 media: listen to pad-removed signals
4031 Listen to the pad-removed signal and remove the stream associated with the
4033 Add signal to be notified of the removed pad.
4034 Remove the fakesink in unprepare()
4035 Fix signatures of the signal methods
4037 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4039 * examples/test-sdp.c:
4040 tests: add example of reusable pipelines
4042 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4044 * gst/rtsp-server/rtsp-stream.c:
4045 * gst/rtsp-server/rtsp-stream.h:
4046 stream: add method to get the srcpad
4048 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4050 * tests/check/gst/media.c:
4051 check: add media prepare/unprepare test
4052 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4054 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4056 * gst/rtsp-server/rtsp-media.c:
4057 media: disconnect from signal handlers in unprepare()
4058 We connected to the pad-added and no-more-pads signals in prepare() so
4059 we need to disconnect from them in unprepare().
4060 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4062 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4064 * gst/rtsp-server/rtsp-media.c:
4065 media: don't free streams array
4066 Don't free the streams array in the unprepare() method, they were not
4068 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4070 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4072 * gst/rtsp-server/rtsp-media.c:
4073 media: don't unref the pipeline in unprepare
4074 Unprepare() should undo what prepare() does. Because the pipeline is
4075 not created in prepare(), we should not unref it in unprepare()
4077 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4079 * gst/rtsp-server/rtsp-stream.c:
4080 stream: clear session and caps for reuse
4081 Set the session and caps to NULL after unref otherwise we might unref
4083 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4085 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4087 * gst/rtsp-server/rtsp-client.c:
4088 client: send out teardown signal before tearing down
4089 The advantage is that in the signal handler you get direct access to
4090 information about what streams are about to get torn down (in the
4091 GstRTSPClientState).
4092 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4094 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4096 * gst/rtsp-server/rtsp-client.c:
4097 * gst/rtsp-server/rtsp-client.h:
4098 client: expose connection
4099 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4101 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4104 Automatic update of common submodule
4105 From aed87ae to 3cb3d3c
4107 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4109 * gst/rtsp-server/rtsp-media.c:
4110 * gst/rtsp-server/rtsp-media.h:
4111 * gst/rtsp-server/rtsp-session-media.c:
4112 * gst/rtsp-server/rtsp-session-media.h:
4113 media: add method to get the base_time of the pipeline
4114 Together with a shared clock, this base-time could eventually be sent to
4115 the client so that it can reconstruct the exact running-time of the clock
4118 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4120 * gst/rtsp-server/Makefile.am:
4121 * gst/rtsp-server/rtsp-media.c:
4122 * gst/rtsp-server/rtsp-media.h:
4123 * gst/rtsp-server/rtsp-sdp.c:
4124 media: add GstNetTimeProvider support
4125 Add a property to let the media provide a GstNetTimeProvider for its clock.
4126 Make methods to get the clock and nettimeprovider
4127 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4128 provider and also the current time of the clock. This should make it possible
4129 for (GStreamer) clients to slave their clock to the server clock.
4131 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4134 Automatic update of common submodule
4135 From 04c7a1e to aed87ae
4137 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4139 * gst/rtsp-server/rtsp-media.c:
4140 media: wait for buffering to complete
4141 Wait for buffering to complete before changing the state to the target state.
4143 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4145 * gst/rtsp-server/rtsp-media.c:
4146 media: small cleanup
4148 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4150 * tests/check/gst/rtspserver.c:
4151 tests: remove extra unref in test_setup_non_existing_stream
4152 The unref is not needed anymore, teardown runs without it.
4153 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4155 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4157 * tests/check/gst/rtspserver.c:
4158 tests: GSocketService cleanup in test_bind_already_in_use
4159 Use g_socket_service_stop so the rtspserver test stops listening for
4160 incoming connections in test_bind_already_in_use.
4161 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4163 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4165 * gst/rtsp-server/rtsp-media-factory.c:
4166 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4167 Instead use a GWeakRef which is safe to use
4168 This is a known GLib bug, see:
4169 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4171 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4173 * gst/rtsp-server/rtsp-client.c:
4174 * gst/rtsp-server/rtsp-media.c:
4175 * gst/rtsp-server/rtsp-media.h:
4176 * gst/rtsp-server/rtsp-sdp.c:
4177 * tests/check/gst/media.c:
4178 * tests/check/gst/rtspserver.c:
4179 rtsp-media/client: Reply to PLAY request with same type of Range
4180 Remember the type of Range from the PLAY request and use the same type for
4183 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4185 * gst/rtsp-server/rtsp-client.c:
4186 * gst/rtsp-server/rtsp-client.h:
4187 * tests/check/gst/client.c:
4188 rtsp-client: expose uri
4190 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4192 * tests/check/gst/mediafactory.c:
4193 tests: Hold ref while creating second media
4194 To test if the media aren't shared, make sure we keep the first one while creating a second
4195 otherwise the same memory address may be reused.
4197 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4200 configure: remove out-of-date comment
4202 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4205 .gitignore: ignore more build files
4207 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4209 * tests/check/Makefile.am:
4210 tests: use right _LIBS variable for gst-plugins-base libs
4212 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4214 * tests/check/Makefile.am:
4215 check: add librtp to libs
4217 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4219 * tests/check/gst/rtspserver.c:
4220 tests: Add test to check selecting a port the server will send from
4222 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4224 * tests/check/gst/rtspserver.c:
4225 tests: Make sure packets are actually received
4227 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4229 * gst/rtsp-server/rtsp-stream.c:
4230 stream: Select unicast address from pool if appropriate
4232 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4234 * gst/rtsp-server/rtsp-stream.c:
4235 stream: Properties are always there in Gst 1.0
4237 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4239 * tests/check/gst/addresspool.c:
4240 tests: Add tests for unicast addresses in pool
4242 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4244 * gst/rtsp-server/rtsp-address-pool.c:
4245 * tests/check/gst/addresspool.c:
4246 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4248 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4250 * docs/libs/gst-rtsp-server-sections.txt:
4251 * gst/rtsp-server/rtsp-address-pool.c:
4252 * gst/rtsp-server/rtsp-address-pool.h:
4253 * gst/rtsp-server/rtsp-stream.c:
4254 * tests/check/gst/addresspool.c:
4255 address-pool: Add unicast addresses
4257 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4260 * gst/rtsp-server/rtsp-server.c:
4261 * tests/check/gst/rtspserver.c:
4262 rtsp-server: Limit the number of threads per server instance
4263 If we exceed the maximum, just round robin the clients over the existing
4266 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4268 * gst/rtsp-server/rtsp-server.c:
4269 rtsp-server: No need to store the GMainContext in the client context
4271 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4273 * tests/check/gst/rtspserver.c:
4274 tests: Add test for client disconnection
4276 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4278 * tests/check/gst/rtspserver.c:
4279 tests: Test client and session timeouts with multiple threads
4281 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4283 * gst/rtsp-server/rtsp-address-pool.c:
4284 * gst/rtsp-server/rtsp-auth.c:
4285 * gst/rtsp-server/rtsp-client.c:
4286 * gst/rtsp-server/rtsp-media-factory-uri.c:
4287 * gst/rtsp-server/rtsp-media-factory.c:
4288 * gst/rtsp-server/rtsp-media.c:
4289 * gst/rtsp-server/rtsp-mount-points.c:
4290 * gst/rtsp-server/rtsp-server.c:
4291 * gst/rtsp-server/rtsp-session-media.c:
4292 * gst/rtsp-server/rtsp-session-pool.c:
4293 * gst/rtsp-server/rtsp-session.c:
4294 Document locking and its order
4296 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4298 * tests/check/gst/rtspserver.c:
4299 tests: Test that slow DESCRIBE don't block other clients
4301 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4303 * tests/check/gst/client.c:
4304 tests: Add tests for client-requested multicast address
4306 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4308 * docs/libs/gst-rtsp-server-sections.txt:
4309 docs: Put the various functions in the right sections
4311 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4313 * docs/libs/gst-rtsp-server-docs.sgml:
4314 * docs/libs/gst-rtsp-server-sections.txt:
4315 * gst/rtsp-server/rtsp-address-pool.c:
4316 * gst/rtsp-server/rtsp-address-pool.h:
4317 docs: Generate docs for GstRTSPAddressPool
4319 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4321 * gst/rtsp-server/rtsp-client.c:
4322 * gst/rtsp-server/rtsp-stream.c:
4323 * gst/rtsp-server/rtsp-stream.h:
4324 client: Check client provided addresses against the address pool
4326 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4328 * gst/rtsp-server/rtsp-address-pool.c:
4329 * gst/rtsp-server/rtsp-address-pool.h:
4330 * tests/check/gst/addresspool.c:
4331 address-pool: Add API to request a specific address from the pool
4332 Also add relevant unit tests.
4334 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4336 * tests/check/gst/mediafactory.c:
4337 tests: Check the passing around of a RTSPAddressPool
4338 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4339 way down to the stream.
4341 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4343 * tests/check/gst/addresspool.c:
4344 tests: Add more tests for the address pool
4346 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4348 * gst/rtsp-server/rtsp-address-pool.c:
4349 address-pool: Fix off by one error
4350 When splitting a port range, the port after a skip is not part of range.
4352 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4355 Automatic update of common submodule
4356 From 2de221c to 04c7a1e
4358 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4361 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4362 AM_CONFIG_HEADER was removed in automake 1.13
4363 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4365 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4368 Automatic update of common submodule
4369 From a942293 to 2de221c
4371 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4373 * gst/rtsp-server/rtsp-client.c:
4374 client: make sure the watch exists while sending data
4375 Protect the send_func with a lock. This allows us to wait for sending
4376 to complete before changing the send_func and user_data. We add an
4377 extra ref to the watch to make sure that it remains valid during
4379 When closing the connection, set the send_func to NULL
4380 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4382 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4384 * tests/check/Makefile.am:
4385 tests: use GST_*_1_0 environment variables everywhere
4386 The _1_0 suffixed environment variables override the
4387 non-suffixed ones, so if we're in an environment that
4388 sets the _1_0 suffixed ones, such as jhbuild, we need
4389 to set those to make sure ours actually always get
4392 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4395 Automatic update of common submodule
4396 From acb04d9 to a942293
4398 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4400 * gst/rtsp-server/rtsp-client.c:
4401 rtsp-client: set the client backlog
4402 Set the client backlog to a reasonable default
4404 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4406 * gst/rtsp-server/rtsp-media.c:
4407 rtsp-media: Make the element a constructor parameter
4408 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4410 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4412 * docs/libs/Makefile.am:
4413 docs: Link with gcov library when gcov is enabled
4414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4416 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4418 * gst/rtsp-server/rtsp-media.c:
4419 media: match prepare with unprepare
4420 Really unprepare when there were an equal amount of prepare calls.
4422 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4424 * gst/rtsp-server/rtsp-media.c:
4425 media: media has to be unprepared in finalize
4426 Because unprepare takes away the last ref on the media.
4428 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4430 * gst/rtsp-server/rtsp-client.c:
4431 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4432 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4433 We can't use the refcount to trigger unprepare because it is the unprepare call
4434 that removes the last refcount after all messages are consumed. What we should
4435 probably do is make a prepared refcount and only unprepare when the refcount
4438 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4440 * gst/rtsp-server/rtsp-media.c:
4441 media: let the source unref the last media ref
4442 the last ref to the media is held by the source so we don't need to add more ref
4443 and unrefs, we simply destroy the media when the source is gone.
4445 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4447 * gst/rtsp-server/rtsp-media.c:
4448 media: improve debug
4450 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4452 * gst/rtsp-server/rtsp-media.c:
4454 Make sure we are in the right state when collecting the position and duration.
4455 Only make ourselves PREPARED when we were previously PREPARING.
4457 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4459 * gst/rtsp-server/rtsp-media.c:
4460 media: use g_object_ref/unref for GObjects
4462 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4464 * gst/rtsp-server/rtsp-client.c:
4465 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4466 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4467 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4468 isn't being used anymore.
4470 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4472 * gst/rtsp-server/rtsp-media.c:
4473 Fix compiler warning
4475 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4477 * gst/rtsp-server/rtsp-media-factory-uri.c:
4478 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4480 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4482 * gst/rtsp-server/rtsp-session-media.h:
4485 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4487 * gst/rtsp-server/rtsp-media.c:
4488 * tests/check/gst/media.c:
4489 media: avoid element leak
4491 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4493 * gst/rtsp-server/rtsp-media.c:
4494 media: require an element in media constructor
4496 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4498 * gst/rtsp-server/rtsp-client.c:
4499 Revert "client: TEARDOWN brings that state to Init again"
4500 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4501 The object is already disposed, there is no point in setting the state.
4503 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4505 * gst/rtsp-server/rtsp-client.c:
4506 client: TEARDOWN brings that state to Init again
4508 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4510 * docs/libs/gst-rtsp-server-sections.txt:
4511 * examples/test-auth.c:
4512 * gst/rtsp-server/rtsp-auth.c:
4513 * gst/rtsp-server/rtsp-auth.h:
4514 * gst/rtsp-server/rtsp-client.c:
4515 * gst/rtsp-server/rtsp-client.h:
4516 * gst/rtsp-server/rtsp-media-factory-uri.c:
4517 * gst/rtsp-server/rtsp-media-factory-uri.h:
4518 * gst/rtsp-server/rtsp-media-factory.c:
4519 * gst/rtsp-server/rtsp-media-factory.h:
4520 * gst/rtsp-server/rtsp-media.c:
4521 * gst/rtsp-server/rtsp-media.h:
4522 * gst/rtsp-server/rtsp-mount-points.c:
4523 * gst/rtsp-server/rtsp-mount-points.h:
4524 * gst/rtsp-server/rtsp-sdp.c:
4525 * gst/rtsp-server/rtsp-server.c:
4526 * gst/rtsp-server/rtsp-server.h:
4527 * gst/rtsp-server/rtsp-session-media.c:
4528 * gst/rtsp-server/rtsp-session-media.h:
4529 * gst/rtsp-server/rtsp-session-pool.c:
4530 * gst/rtsp-server/rtsp-session-pool.h:
4531 * gst/rtsp-server/rtsp-session.c:
4532 * gst/rtsp-server/rtsp-session.h:
4533 * gst/rtsp-server/rtsp-stream-transport.c:
4534 * gst/rtsp-server/rtsp-stream-transport.h:
4535 * gst/rtsp-server/rtsp-stream.c:
4536 * gst/rtsp-server/rtsp-stream.h:
4537 * tests/check/gst/media.c:
4538 rtsp: make object details private
4539 Make all object details private
4540 Add methods to access private bits
4542 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4544 * tests/check/Makefile.am:
4545 * tests/check/gst/media.c:
4546 tests: add media tests
4548 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4550 * gst/rtsp-server/rtsp-media.c:
4551 media: check if prepared for some methods
4552 Check that the media object is prepared before doing seek and getting the
4553 current position etc.
4554 Add some g_return checks.
4556 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4558 * tests/check/Makefile.am:
4559 * tests/check/gst/mediafactory.c:
4560 tests: add mediafactory test
4562 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4564 * gst/rtsp-server/rtsp-stream.c:
4565 stream: improve debug
4567 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4569 * gst/rtsp-server/rtsp-media.c:
4570 * gst/rtsp-server/rtsp-media.h:
4571 media: unref pipeline in finalize to avoid leaking it
4573 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4575 * gst/rtsp-server/rtsp-media-factory-uri.c:
4576 * gst/rtsp-server/rtsp-media.c:
4577 rtsp: use gst_object_unref on GstObjects
4579 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4581 * gst/rtsp-server/rtsp-media-factory.c:
4582 media-factory: require an url
4584 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4586 * examples/test-uri.c:
4587 examples: fix include
4589 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4591 * gst/rtsp-server/rtsp-server.h:
4592 server: remove unused include
4594 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4596 * tests/check/Makefile.am:
4597 * tests/check/gst/mountpoints.c:
4598 tests: add test for mountpoints
4600 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4602 * gst/rtsp-server/rtsp-client.c:
4603 client: fix factory leak
4604 Keep the factory in the state object only for authorization checks and make
4605 sure we unref it on failure. Also don't keep invalid objects in the state
4608 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4610 * gst/rtsp-server/rtsp-mount-points.c:
4611 mounts: add g_return_if guards
4613 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4615 * tests/check/gst/client.c:
4616 tests: add more tests
4618 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4620 * gst/rtsp-server/rtsp-client.c:
4621 client: improve debug
4623 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4625 * gst/rtsp-server/rtsp-client.c:
4626 client: improve debug and fix leaks
4627 Cleanup the uri and session when there is a bad request.
4629 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4634 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4636 * tests/check/gst/client.c:
4637 test: add test for session in options request
4639 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4641 * gst/rtsp-server/rtsp-client.c:
4642 client: use 454 when session can't be found
4643 We should use 454 when a session can't be found because there was no session
4644 pool configured in the server. This is not a server configuration problem
4645 because the server on which the request is done might not be the same one that
4646 will keep the sessions for us and so it does not need to support sessions.
4648 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4650 * gst/rtsp-server/rtsp-client.c:
4651 client: only free connection when there is one
4652 It's possible that the client doesn't have a connection when we try to free it.
4654 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * tests/check/Makefile.am:
4657 * tests/check/gst/client.c:
4658 tests: add unit test for the client object
4660 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4662 * gst/rtsp-server/rtsp-client.c:
4663 client: small cleanup
4665 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4667 * gst/rtsp-server/rtsp-client.h:
4668 client: remove unused include
4670 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4672 * gst/rtsp-server/rtsp-client.c:
4673 client: fix compilation
4675 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4677 * gst/rtsp-server/rtsp-client.c:
4678 client: call destroy without the lock
4680 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4682 * gst/rtsp-server/rtsp-client.c:
4683 * gst/rtsp-server/rtsp-client.h:
4684 client: make the client usable without a socket
4685 Make a method to let the client handle a message and a callback when the client
4686 wants us to send a response message back. This makes it possible to also use the
4687 client object without the sockets, which should make it easier to test.
4689 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4691 * gst/rtsp-server/rtsp-client.c:
4692 * gst/rtsp-server/rtsp-client.h:
4693 client: small cleanup
4695 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4697 * docs/libs/gst-rtsp-server-sections.txt:
4698 * gst/rtsp-server/rtsp-client.c:
4699 * gst/rtsp-server/rtsp-client.h:
4700 * gst/rtsp-server/rtsp-server.c:
4701 client: remove reference to server
4702 We don't need to keep a ref to the server
4704 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4706 * gst/rtsp-server/rtsp-client.c:
4707 * gst/rtsp-server/rtsp-client.h:
4709 Also add some g_return_if()
4711 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4713 * gst/rtsp-server/rtsp-client.c:
4714 client: log more errors
4716 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4718 * gst/rtsp-server/rtsp-client.c:
4719 client: fix compilation
4721 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4723 * gst/rtsp-server/rtsp-client.c:
4724 * gst/rtsp-server/rtsp-client.h:
4725 client: add generic close-after-send support
4726 Add a property to send_response() to close the connection after the response has
4727 been sent to the client.
4729 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4732 * docs/libs/gst-rtsp-server-docs.sgml:
4733 * docs/libs/gst-rtsp-server-sections.txt:
4734 * docs/libs/gst-rtsp-server.types:
4735 * examples/test-auth.c:
4736 * examples/test-launch.c:
4737 * examples/test-mp4.c:
4738 * examples/test-multicast.c:
4739 * examples/test-multicast2.c:
4740 * examples/test-ogg.c:
4741 * examples/test-readme.c:
4742 * examples/test-sdp.c:
4743 * examples/test-uri.c:
4744 * examples/test-video.c:
4745 * gst/rtsp-server/Makefile.am:
4746 * gst/rtsp-server/rtsp-auth.h:
4747 * gst/rtsp-server/rtsp-client.c:
4748 * gst/rtsp-server/rtsp-client.h:
4749 * gst/rtsp-server/rtsp-media-mapping.c:
4750 * gst/rtsp-server/rtsp-media-mapping.h:
4751 * gst/rtsp-server/rtsp-mount-points.c:
4752 * gst/rtsp-server/rtsp-mount-points.h:
4753 * gst/rtsp-server/rtsp-server.c:
4754 * gst/rtsp-server/rtsp-server.h:
4755 * gst/rtsp-server/rtsp-session-media.c:
4756 * gst/rtsp-server/rtsp-session-pool.c:
4757 * gst/rtsp-server/rtsp-session-pool.h:
4758 * tests/check/gst/rtspserver.c:
4759 MediaMapping -> MountPoints
4760 Describes better what the object manages.
4762 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4765 configure: bump required version of -base
4767 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4769 * gst/rtsp-server/rtsp-media.c:
4772 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4774 * gst/rtsp-server/rtsp-media.c:
4775 * gst/rtsp-server/rtsp-media.h:
4776 media: support more Range formats
4777 Use the new -base methods to convert the Range string into a seek start and stop
4780 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4782 * examples/test-launch.c:
4783 examples: fix whitespace
4785 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4787 * examples/test-auth.c:
4788 test-auth: add example of how to remove sessions
4789 Add an example of the session filter api.
4791 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4793 * examples/test-uri.c:
4794 test-uri: remove mapping example
4796 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4798 * examples/test-uri.c:
4799 test-uri: fix callback signature
4801 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * gst/rtsp-server/rtsp-media-factory.c:
4804 factory: keep ref to factory while media active
4805 While the media from a factory is alive, keep a ref to the factory.
4806 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4808 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * gst/rtsp-server/rtsp-media-factory-uri.c:
4811 factory-uri: add some debug
4813 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4815 * gst/rtsp-server/rtsp-stream.c:
4816 stream: set udp sources to PLAYING
4817 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4818 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4820 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4822 * gst/rtsp-server/rtsp-media-factory-uri.c:
4823 factory-uri: take ref to factory
4824 Take a ref to the factory that we place in our list.
4826 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4828 * tests/Makefile.am:
4829 * tests/test-reuse.c:
4830 test: add test for server reuse
4831 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4833 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4835 * gst/rtsp-server/rtsp-server.c:
4836 server: start and stop multiple times
4837 Stop listening on the RTSP port when the GSource is removed, so clients
4838 can't connect and the server can be started again.
4839 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4841 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4843 * gst/rtsp-server/rtsp-server.c:
4844 server: fix small leak
4846 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4848 * gst/rtsp-server/rtsp-media.c:
4849 media: unref source in finish_unprepare
4850 The source is created in prepare, unref it in finish_unprepare.
4851 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4853 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4855 * gst/rtsp-server/rtsp-client.c:
4856 * gst/rtsp-server/rtsp-media.c:
4857 rtsp-media: remove bus watch before finalizing
4858 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4859 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4860 the GDestroyNotify function.
4861 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4862 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4863 gst_rtsp_media_unprepare before unreffing the media.
4864 This way, the bus watch will be removed before the media is finalized.
4865 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4867 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4869 * gst/rtsp-server/rtsp-client.c:
4870 * gst/rtsp-server/rtsp-client.h:
4871 client: wait until the TEARDOWN response is sent to close the connection
4872 Responses can be sent async so we need to wait until the TEARDOWN response has
4873 been written before we close the connection to the client. This avoids the risk
4874 of writing/polling closed sockets.
4875 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4877 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4879 * gst/rtsp-server/rtsp-stream.c:
4880 rtsp-stream: plug socket leak
4881 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4883 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4886 Automatic update of common submodule
4887 From 6bb6951 to a72faea
4889 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4891 * gst/rtsp-server/rtsp-media-factory-uri.c:
4892 rtsp-server: don't use deprecated API
4894 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4896 * gst/rtsp-server/rtsp-client.c:
4897 rtsp-client: fix unused-but-set-variable compiler warning
4898 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
4900 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4903 * docs/libs/gst-rtsp-server-sections.txt:
4904 * gst/rtsp-server/rtsp-client.c:
4907 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4909 * examples/Makefile.am:
4910 * examples/test-multicast2.c:
4911 examples: add another multicast example
4912 Add an example for how to configure separate multicast ranges for each media
4915 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4917 * examples/test-multicast.c:
4920 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4922 * gst/rtsp-server/rtsp-client.c:
4923 * gst/rtsp-server/rtsp-media.c:
4924 * gst/rtsp-server/rtsp-session-media.c:
4925 * gst/rtsp-server/rtsp-session-media.h:
4926 * gst/rtsp-server/rtsp-stream-transport.c:
4927 * gst/rtsp-server/rtsp-stream-transport.h:
4928 stream: use the address managed by the stream
4929 Use the address managed by the stream for multicast. This allows us to have 1
4930 multicast address for each stream.
4931 Because the address is now managed by the stream we don't have to pass it around
4933 Set the address pool on the streams.
4935 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4937 * gst/rtsp-server/rtsp-client.c:
4938 * gst/rtsp-server/rtsp-media.c:
4939 * gst/rtsp-server/rtsp-stream.c:
4942 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4944 * gst/rtsp-server/rtsp-media.c:
4945 * gst/rtsp-server/rtsp-media.h:
4946 media: add signal for new streams
4947 This allows applications to listen for new streams and configure properties on
4948 them, like the address pool.
4950 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4952 * gst/rtsp-server/rtsp-media.c:
4953 media: configure address pool in new streams
4955 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4957 * gst/rtsp-server/rtsp-stream.c:
4958 * gst/rtsp-server/rtsp-stream.h:
4959 stream: add methods to deal with address pool
4960 Add methods to get and set the address pool for the stream
4961 Add method to allocate and get the multicast addresses for this stream.
4963 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4965 * docs/libs/gst-rtsp-server-sections.txt:
4966 * gst/rtsp-server/rtsp-media.c:
4967 * gst/rtsp-server/rtsp-media.h:
4968 media: remove MTU property
4969 It is a stream property
4971 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4973 * gst/rtsp-server/rtsp-client.c:
4974 client: set blocksize only on stream
4975 Set the blocksize only on the current stream.
4977 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4979 * gst/rtsp-server/rtsp-stream.c:
4980 stream: share src and sink sockets
4981 the allocated socket is in the used-socket property, not socket.
4983 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4985 * gst/rtsp-server/rtsp-address-pool.c:
4986 * gst/rtsp-server/rtsp-address-pool.h:
4987 * gst/rtsp-server/rtsp-client.c:
4988 * gst/rtsp-server/rtsp-session-media.c:
4989 * gst/rtsp-server/rtsp-session-media.h:
4990 * gst/rtsp-server/rtsp-stream-transport.c:
4991 * gst/rtsp-server/rtsp-stream-transport.h:
4992 * tests/check/gst/addresspool.c:
4993 rtsp: make address-pool return an address object
4994 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
4995 store more info in the structure and allows us to more easily return the address
4996 to the right pool when no longer needed.
4997 Pass the address to the StreamTransport so that we can return it to the pool
4998 when the stream transport is freed or changed.
5000 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5002 * examples/Makefile.am:
5003 * examples/test-multicast.c:
5004 examples: add multicast example
5005 Show how to set up the multicast address pool so that media can be
5006 server with multicast.
5008 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5010 * gst/rtsp-server/rtsp-client.c:
5011 * gst/rtsp-server/rtsp-media-factory.c:
5012 * gst/rtsp-server/rtsp-media-factory.h:
5013 * gst/rtsp-server/rtsp-media.c:
5014 * gst/rtsp-server/rtsp-media.h:
5015 rtsp: use AddressPool
5016 Remove the multicast_group property.
5017 Use the configured addresspool to allocate multicast addresses.
5019 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5021 * gst/rtsp-server/rtsp-address-pool.c:
5022 * gst/rtsp-server/rtsp-address-pool.h:
5023 address-pool: add clear method
5025 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5027 * gst/rtsp-server/rtsp-address-pool.c:
5028 address-pool: small cleanups
5030 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5032 * tests/check/Makefile.am:
5033 * tests/check/gst/addresspool.c:
5034 tests: add addresspool unit test
5036 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5038 * gst/rtsp-server/Makefile.am:
5039 * gst/rtsp-server/rtsp-address-pool.c:
5040 * gst/rtsp-server/rtsp-address-pool.h:
5041 address-pool: add object to manage multicast addresses
5042 Make an object that can manage a rage of multicast addresses and ports.
5044 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5046 * gst/rtsp-server/rtsp-server.c:
5047 server: set default max-threads property
5049 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5051 * gst/rtsp-server/rtsp-media.c:
5052 media: wait for concurrent _prepare
5053 If a prepare is busy, wait for the result.
5055 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5057 * gst/rtsp-server/rtsp-media.c:
5058 media: add lock around message handler
5059 We don't want to dispatch messages while we are still processing the result of
5062 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5064 * gst/rtsp-server/rtsp-media.c:
5065 * gst/rtsp-server/rtsp-media.h:
5066 media: add lock to protect state changes
5068 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5070 * gst/rtsp-server/rtsp-stream.c:
5071 * gst/rtsp-server/rtsp-stream.h:
5074 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5076 * gst/rtsp-server/rtsp-stream-transport.c:
5077 * gst/rtsp-server/rtsp-stream-transport.h:
5078 * gst/rtsp-server/rtsp-stream.c:
5079 stream-transport: add keep-alive method
5081 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5083 * gst/rtsp-server/rtsp-stream-transport.c:
5084 * gst/rtsp-server/rtsp-stream-transport.h:
5085 * gst/rtsp-server/rtsp-stream.c:
5086 stream-transport: add method to handle RTP/RTCP
5087 Call new methods instead of poking into the structures directly.
5089 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5091 * gst/rtsp-server/rtsp-session-media.c:
5092 * gst/rtsp-server/rtsp-session-media.h:
5093 session-media: add locking
5095 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5097 * gst/rtsp-server/rtsp-session.c:
5098 * gst/rtsp-server/rtsp-session.h:
5099 session: add locking
5101 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5103 * gst/rtsp-server/rtsp-server.c:
5104 server: free old socket
5106 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5108 * gst/rtsp-server/rtsp-media-mapping.c:
5109 * gst/rtsp-server/rtsp-media-mapping.h:
5110 mapping: add locking
5112 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5114 * gst/rtsp-server/rtsp-media-factory.c:
5115 media-factory: add locking
5117 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5119 * gst/rtsp-server/rtsp-auth.c:
5120 * gst/rtsp-server/rtsp-auth.h:
5123 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5125 * gst/rtsp-server/rtsp-server.c:
5126 * gst/rtsp-server/rtsp-server.h:
5127 server: add max-thread property
5129 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5131 * gst/rtsp-server/rtsp-server.c:
5132 * gst/rtsp-server/rtsp-server.h:
5133 server: use a threadpool for the mainloops
5135 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5137 * gst/rtsp-server/rtsp-client.c:
5138 * gst/rtsp-server/rtsp-client.h:
5139 client: rename method
5140 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5141 don't really create the client from the socket, we use the socket for the
5144 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5146 * gst/rtsp-server/rtsp-client.c:
5147 * gst/rtsp-server/rtsp-client.h:
5148 * gst/rtsp-server/rtsp-server.c:
5149 server: rework maincontext handling in clients
5150 Make a separate method to attach a client to a MainContext.
5151 Let the server decide in what GMainContext the client will operate and give this
5152 context to the client in attach. Then the server can later decide to use a
5153 separate thread for each client or just use the mainthread.
5155 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5157 * gst/rtsp-server/rtsp-client.c:
5158 * gst/rtsp-server/rtsp-session.c:
5159 * gst/rtsp-server/rtsp-session.h:
5160 session: move session header code in session object
5162 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5166 * examples/test-auth.c:
5167 * examples/test-launch.c:
5168 * examples/test-mp4.c:
5169 * examples/test-ogg.c:
5170 * examples/test-readme.c:
5171 * examples/test-sdp.c:
5172 * examples/test-uri.c:
5173 * examples/test-video.c:
5174 * gst/rtsp-server/rtsp-auth.c:
5175 * gst/rtsp-server/rtsp-auth.h:
5176 * gst/rtsp-server/rtsp-client.c:
5177 * gst/rtsp-server/rtsp-client.h:
5178 * gst/rtsp-server/rtsp-media-factory-uri.c:
5179 * gst/rtsp-server/rtsp-media-factory-uri.h:
5180 * gst/rtsp-server/rtsp-media-factory.c:
5181 * gst/rtsp-server/rtsp-media-factory.h:
5182 * gst/rtsp-server/rtsp-media-mapping.c:
5183 * gst/rtsp-server/rtsp-media-mapping.h:
5184 * gst/rtsp-server/rtsp-media.c:
5185 * gst/rtsp-server/rtsp-media.h:
5186 * gst/rtsp-server/rtsp-params.c:
5187 * gst/rtsp-server/rtsp-params.h:
5188 * gst/rtsp-server/rtsp-sdp.c:
5189 * gst/rtsp-server/rtsp-sdp.h:
5190 * gst/rtsp-server/rtsp-server.c:
5191 * gst/rtsp-server/rtsp-server.h:
5192 * gst/rtsp-server/rtsp-session-media.c:
5193 * gst/rtsp-server/rtsp-session-media.h:
5194 * gst/rtsp-server/rtsp-session-pool.c:
5195 * gst/rtsp-server/rtsp-session-pool.h:
5196 * gst/rtsp-server/rtsp-session.c:
5197 * gst/rtsp-server/rtsp-session.h:
5198 * gst/rtsp-server/rtsp-stream-transport.c:
5199 * gst/rtsp-server/rtsp-stream-transport.h:
5200 * gst/rtsp-server/rtsp-stream.c:
5201 * gst/rtsp-server/rtsp-stream.h:
5202 * tests/check/gst/rtspserver.c:
5203 * tests/test-cleanup.c:
5206 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5208 * gst/rtsp-server/rtsp-media.c:
5209 * gst/rtsp-server/rtsp-session-media.c:
5210 * gst/rtsp-server/rtsp-session.c:
5211 rtsp-server: added annotations to indicate type of ownership transfer of return values
5212 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5214 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5217 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5219 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5222 * bindings/Makefile.am:
5223 * bindings/vala/Makefile.am:
5224 * bindings/vala/gst-rtsp-server-0.10.deps:
5225 * bindings/vala/gst-rtsp-server-0.10.vapi:
5226 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5227 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5228 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5229 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5230 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5232 bindings: remove vala bindings
5233 They'll be reunited with the other GStreamer bindings
5234 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5236 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5238 * gst/rtsp-server/rtsp-client.c:
5239 * gst/rtsp-server/rtsp-session-media.c:
5240 * gst/rtsp-server/rtsp-session-media.h:
5241 * gst/rtsp-server/rtsp-stream-transport.c:
5242 * gst/rtsp-server/rtsp-stream-transport.h:
5243 rtsp: only create transport when needed
5244 Only create the StreamTransport when configured.
5246 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5248 * gst/rtsp-server/rtsp-client.c:
5249 client: small cleanup
5251 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5253 * gst/rtsp-server/rtsp-client.c:
5254 * gst/rtsp-server/rtsp-client.h:
5255 * gst/rtsp-server/rtsp-stream-transport.c:
5256 * gst/rtsp-server/rtsp-stream-transport.h:
5257 rtsp: refactor configuration of transport
5258 Move the configuration of the transport to a place where it makes
5261 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5263 * gst/rtsp-server/rtsp-client.c:
5264 client: refactor transport parsing
5266 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5268 * gst/rtsp-server/rtsp-client.c:
5269 client: refuse to change the MTU on shared media
5270 If we change the MTU of chared media, it changes for all clients.
5271 We don't want to set the MTU to something large for clients that
5274 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5276 * examples/test-mp4.c:
5277 * gst/rtsp-server/rtsp-media.c:
5278 small fixes to docs and debug
5280 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5282 * gst/rtsp-server/rtsp-stream.c:
5283 stream: transports must already have been removed
5285 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5287 * gst/rtsp-server/rtsp-media.c:
5288 * gst/rtsp-server/rtsp-stream.c:
5289 * gst/rtsp-server/rtsp-stream.h:
5290 stream: improve join and leave of the pipeline
5292 Do the cleanup properly
5295 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5297 * gst/rtsp-server/rtsp-media.c:
5298 media: move unprepare below default implementation
5299 Makes it easier to find the default implementation
5301 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5303 * gst/rtsp-server/rtsp-media.c:
5304 media: signal unprepared when we actually finish
5306 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5308 * gst/rtsp-server/rtsp-media.c:
5309 media: no need to unlock, unprepare does that when needed
5311 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5313 * docs/libs/gst-rtsp-server-sections.txt:
5314 * gst/rtsp-server/rtsp-media-factory.h:
5315 * gst/rtsp-server/rtsp-media-mapping.c:
5316 * gst/rtsp-server/rtsp-media.h:
5317 * gst/rtsp-server/rtsp-params.c:
5318 * gst/rtsp-server/rtsp-server.c:
5319 * gst/rtsp-server/rtsp-session-pool.h:
5320 * gst/rtsp-server/rtsp-session.c:
5321 * gst/rtsp-server/rtsp-session.h:
5322 * gst/rtsp-server/rtsp-stream-transport.h:
5323 * gst/rtsp-server/rtsp-stream.h:
5326 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5328 * gst/rtsp-server/rtsp-client.c:
5329 * gst/rtsp-server/rtsp-media-mapping.h:
5330 * gst/rtsp-server/rtsp-media.c:
5331 * gst/rtsp-server/rtsp-media.h:
5332 * gst/rtsp-server/rtsp-server.h:
5333 * gst/rtsp-server/rtsp-stream.c:
5334 * gst/rtsp-server/rtsp-stream.h:
5335 rtsp: fix MTU setting
5336 Fix setting of the MTU. There is no need for a vmethod.
5338 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5343 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5346 configure: bump version number after refactoring
5348 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5350 * gst/rtsp-server/Makefile.am:
5351 * gst/rtsp-server/rtsp-client.c:
5352 * gst/rtsp-server/rtsp-client.h:
5353 * gst/rtsp-server/rtsp-media-factory-uri.c:
5354 * gst/rtsp-server/rtsp-media-factory.c:
5355 * gst/rtsp-server/rtsp-media-factory.h:
5356 * gst/rtsp-server/rtsp-media.c:
5357 * gst/rtsp-server/rtsp-media.h:
5358 * gst/rtsp-server/rtsp-sdp.c:
5359 * gst/rtsp-server/rtsp-session-media.c:
5360 * gst/rtsp-server/rtsp-session-media.h:
5361 * gst/rtsp-server/rtsp-session.c:
5362 * gst/rtsp-server/rtsp-session.h:
5363 * gst/rtsp-server/rtsp-stream-transport.c:
5364 * gst/rtsp-server/rtsp-stream-transport.h:
5365 * gst/rtsp-server/rtsp-stream.c:
5366 * gst/rtsp-server/rtsp-stream.h:
5367 rtsp: massive refactoring
5368 Make GObjects from the remaining simple structures.
5369 Remove GstRTSPSessionStream, it's not needed.
5370 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5371 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5372 a GstRTSPStream should be transported to a client.
5373 Rename GstRTSPMediaFactory::get_element -> create_element because that
5374 more accurately describes what it does.
5375 Make nice methods instead of poking in the structures.
5376 Move some methods inside the relevant object source code.
5377 Use GPtrArray to store objects instead of plain arrays, it is more
5378 natural and allows us to more easily clean up.
5379 Move the allocation of udp ports to the Stream object. The Stream object
5380 contains the elements needed to stream the media to a client.
5381 Improve the prepare and unprepare methods. Unprepare should now undo
5382 everything prepare did. Improve also async unprepare when doing EOS on
5383 shutdown. Make sure we always unprepare correctly.
5385 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5387 * gst/rtsp-server/rtsp-client.c:
5388 rtsp-client: Unref server address clients connected to
5389 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5391 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5393 * gst/rtsp-server/rtsp-server.c:
5394 rtsp-server: don't ref server socket if it is NULL
5395 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5396 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5398 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5400 * tests/check/Makefile.am:
5401 tests: Add libgio link dependency
5402 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5404 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5406 * gst/rtsp-server/rtsp-media-mapping.c:
5407 * gst/rtsp-server/rtsp-media-mapping.h:
5408 rtsp-media-mapping: rename find_media vfunc to find_factory
5409 The virtual method and class method should have the same name
5410 so it is correctly represented in GIR file
5411 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5413 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5415 * gst/rtsp-server/rtsp-auth.c:
5416 * gst/rtsp-server/rtsp-client.c:
5417 * gst/rtsp-server/rtsp-media-factory-uri.c:
5418 * gst/rtsp-server/rtsp-media-factory.c:
5419 * gst/rtsp-server/rtsp-media-mapping.c:
5420 * gst/rtsp-server/rtsp-media.c:
5421 * gst/rtsp-server/rtsp-server.c:
5422 * gst/rtsp-server/rtsp-session-pool.c:
5423 * gst/rtsp-server/rtsp-session.c:
5424 rtsp-server: fixed comments and GIR annotations
5425 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5427 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5429 * gst/rtsp-server/rtsp-media-mapping.c:
5430 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5432 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5434 * gst/rtsp-server/rtsp-server.c:
5435 rtsp-server: allow binding on port 0 (binds on a random port)
5437 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5439 * gst/rtsp-server/rtsp-server.c:
5440 * gst/rtsp-server/rtsp-server.h:
5441 rtsp-server: add bound-port property
5442 bound-port can be used to retrieve the port number when the server is bound on
5443 port 0, which binds on a random port.
5445 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5447 * gst/rtsp-server/rtsp-media-factory.c:
5448 * gst/rtsp-server/rtsp-media-factory.h:
5449 rtsp-media-factory: make ::get_element overridable by GI bindings
5450 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5451 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5452 as the invoker for ::get_element(), making it overridable by GI generated
5455 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5457 * gst/rtsp-server/rtsp-media-factory-uri.c:
5458 rtsp-media-factory-uri: don't autoplug parsers in a loop
5459 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5462 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5464 * gst/rtsp-server/Makefile.am:
5465 Explicitly link against gio. Fix link error on mac.
5467 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5469 * gst/rtsp-server/rtsp-session.c:
5470 session: add ttl to the transport header in SETUP
5471 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5473 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5475 * gst/rtsp-server/rtsp-client.c:
5476 * gst/rtsp-server/rtsp-client.h:
5477 * gst/rtsp-server/rtsp-media.c:
5478 client: Use client transport settings for multicast if allowed.
5479 This patch makes it possible for the client to send transport settings for
5480 multicast (destination && ttl). Client settings must be explicitly allowed or
5481 the server will use its own settings.
5482 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5484 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5487 Automatic update of common submodule
5488 From 6c0b52c to 6bb6951
5490 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5492 * gst/rtsp-server/rtsp-client.c:
5493 rtsp-client: do not destroy the rtsp watch
5494 Don't destroy the client watch while dispatching. The rtsp watch is
5495 automatically destroyed after the rtsp watch function closed() has
5497 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5499 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5502 Automatic update of common submodule
5503 From 4f962f7 to 6c0b52c
5505 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5507 * gst/rtsp-server/rtsp-media.c:
5508 media: fix check for seekability
5510 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-client.c:
5513 client: use more GIO
5514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5516 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5518 * gst/rtsp-server/rtsp-server.c:
5519 server: remove obsolete includes
5521 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5523 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5524 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5525 be available in "on_new_ssrc". The transports are added in
5526 gst_rtsp_media_set_state when going to PLAYING state. However,
5527 "on_new_ssrc" might be called before this happens.
5528 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5530 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5532 * gst/rtsp-server/rtsp-client.c:
5533 * gst/rtsp-server/rtsp-client.h:
5534 rtsp-client: add signals for rtsp requests (fixes #683287)
5536 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5538 * gst/rtsp-server/rtsp-client.c:
5539 * gst/rtsp-server/rtsp-client.h:
5540 add new-session signal to rtsp-client (fixes #683058)
5542 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5545 Automatic update of common submodule
5546 From 668acee to 4f962f7
5548 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5550 * gst/rtsp-server/rtsp-server.c:
5551 * tests/check/gst/rtspserver.c:
5552 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5553 Do not assume that *error is set in g_socket_address_enumerator_next.
5554 Added test_bind_already_in_use unit-test.
5555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5557 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5560 Automatic update of common submodule
5561 From 94ccf4c to 668acee
5563 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5565 * gst/rtsp-server/rtsp-client.c:
5566 * gst/rtsp-server/rtsp-client.h:
5567 rtsp-client: make create_sdp virtual method
5568 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5570 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5573 Automatic update of common submodule
5574 From 98e386f to 94ccf4c
5576 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5578 * gst/rtsp-server/rtsp-client.c:
5581 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5583 * gst/rtsp-server/rtsp-client.c:
5584 * gst/rtsp-server/rtsp-client.h:
5585 * gst/rtsp-server/rtsp-server.c:
5586 * gst/rtsp-server/rtsp-server.h:
5587 rtsp-server: use an existing socket to establish HTTP tunnel
5588 Make it possible to transfer a socket from an HTTP server to be used as
5589 an RTSP over HTTP tunnel.
5591 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5593 * gst/rtsp-server/rtsp-client.c:
5594 * gst/rtsp-server/rtsp-media.c:
5595 * gst/rtsp-server/rtsp-media.h:
5596 rtsp: Handle the blocksize parameter
5597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5599 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5601 * tests/check/Makefile.am:
5602 * tests/check/gst/rtspserver.c:
5603 Have unit test get header from source dir, not installed dir
5604 This makes compilation of unit tests work in a build directory other
5605 than the source directory.
5606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5608 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5610 * gst/rtsp-server/rtsp-media.c:
5611 rtsp-media: update for gst_element_make_from_uri() changes
5613 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5616 * tests/Makefile.am:
5617 * tests/check/Makefile.am:
5618 * tests/check/gst/rtspserver.c:
5620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5622 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5624 * gst/rtsp-server/rtsp-media.c:
5625 rtsp-media: don't collect media stats when going to NULL
5626 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5628 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5630 * gst/rtsp-server/rtsp-client.c:
5631 client: don't leak transports
5633 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5635 * gst/rtsp-server/rtsp-client.c:
5636 rtsp-client: free transport on no_stream in SETUP handler
5638 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5640 * gst/rtsp-server/rtsp-client.c:
5641 rtsp-client: changed session media iteration
5642 In client_unlink_session: now don't iterate in session->medias
5643 list where items are removed by gst_rtsp_session_release_media.
5644 Instead, repeatedly remove the first item.
5646 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5648 * gst/rtsp-server/rtsp-client.c:
5649 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5650 GstRTSPSessionMedia is not a GObject type. When the
5651 GstRTSPSession is freed, it will free the media.
5653 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5655 * gst/rtsp-server/rtsp-media-factory.c:
5656 factory: plug pad leak in collect_streams
5657 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5658 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5659 will take one reference, and the other reference will otherwise
5662 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5665 configure: suppress some warnings when debug is disabled
5666 Warnings about unused variables should be suppressed if core has the
5667 debug system disabled.
5668 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5670 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5672 * docs/libs/Makefile.am:
5673 docs: fix build in uninstalled setup
5674 Include gst-plugins-base libs properly.
5676 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5678 * docs/libs/gst-rtsp-server.types:
5679 docs: include headers defining rtsp-server object types
5680 Fixes compiler warnings during docs build.
5681 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5683 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5686 configure: Add warning flags for compiler when configuring
5687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5689 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5692 Automatic update of common submodule
5693 From 03a0e57 to 98e386f
5695 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5698 Automatic update of common submodule
5699 From 1fab359 to 03a0e57
5701 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5703 * gst/rtsp-server/rtsp-client.c:
5704 client: fix GSocketAddress leak in gst_rtsp_client_accept
5705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5707 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5710 Automatic update of common submodule
5711 From f1b5a96 to 1fab359
5713 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5716 Automatic update of common submodule
5717 From 92b7266 to f1b5a96
5719 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5722 Automatic update of common submodule
5723 From ec1c4a8 to 92b7266
5725 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5728 Automatic update of common submodule
5729 From 3429ba6 to ec1c4a8
5731 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5733 * gst/rtsp-server/rtsp-auth.c:
5734 * gst/rtsp-server/rtsp-client.c:
5735 * gst/rtsp-server/rtsp-media-factory-uri.c:
5736 * gst/rtsp-server/rtsp-server.c:
5737 rtsp: fix compiler warnings
5738 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5740 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5743 Automatic update of common submodule
5744 From dc70203 to 3429ba6
5746 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5748 * gst/rtsp-server/rtsp-client.c:
5749 * gst/rtsp-server/rtsp-media-factory.c:
5750 * gst/rtsp-server/rtsp-media-factory.h:
5751 * gst/rtsp-server/rtsp-media.c:
5752 * gst/rtsp-server/rtsp-media.h:
5753 * gst/rtsp-server/rtsp-server.c:
5754 * gst/rtsp-server/rtsp-server.h:
5755 * gst/rtsp-server/rtsp-session-pool.c:
5756 * gst/rtsp-server/rtsp-session-pool.h:
5757 rtsp-server: port to new thread API
5759 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5762 Automatic update of common submodule
5763 From 6db25be to dc70203
5765 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5767 * gst/rtsp-server/rtsp-auth.c:
5768 * gst/rtsp-server/rtsp-auth.h:
5769 * gst/rtsp-server/rtsp-client.c:
5770 rtsp-server: Fix compilation and compiler warnings
5772 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5776 * gst/rtsp-server/Makefile.am:
5777 configure: Modernize autotools setup a bit
5778 Also we now only create tar.bz2 and tar.xz tarballs.
5780 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5783 Automatic update of common submodule
5784 From 464fe15 to 6db25be
5786 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5789 Automatic update of common submodule
5790 From 7fda524 to 464fe15
5792 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5795 * docs/libs/Makefile.am:
5796 * docs/version.entities.in:
5798 * gst/rtsp-server/Makefile.am:
5799 * pkgconfig/Makefile.am:
5800 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5801 * pkgconfig/gstreamer-rtsp-server.pc.in:
5802 * tests/Makefile.am:
5803 rtsp-server: Update versioning
5805 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5807 Merge remote-tracking branch 'origin/0.10'
5809 gst/rtsp-server/rtsp-session-pool.c
5811 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5813 * gst/rtsp-server/rtsp-session-pool.c:
5814 rtsp-server: Don't use deprecated GLib API
5816 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5818 Replace master with 0.11
5820 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5822 Merge branch 'master' into 0.11
5824 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5826 Merge branch 'master' into 0.11
5828 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5831 A couple minor typo fixes
5833 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5835 * gst/rtsp-server/rtsp-media.c:
5836 media: fix state of the appqueue
5838 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5840 * gst/rtsp-server/rtsp-media-factory-uri.c:
5841 factory: use videoconvert
5843 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5845 * gst/rtsp-server/rtsp-media-factory-uri.c:
5846 factory: change to new style caps
5848 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5850 * gst/rtsp-server/rtsp-client.c:
5851 * gst/rtsp-server/rtsp-client.h:
5852 * gst/rtsp-server/rtsp-media-factory-uri.c:
5853 * gst/rtsp-server/rtsp-media.c:
5854 * gst/rtsp-server/rtsp-server.c:
5855 * gst/rtsp-server/rtsp-server.h:
5856 * gst/rtsp-server/rtsp-session-pool.c:
5857 rtsp-server: port to GIO
5860 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5863 configure: fix build
5865 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5868 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5869 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5871 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5874 * examples/Makefile.am:
5875 First rule of gst-rtsp-server club: don't talk about gst-phonon
5877 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5880 * pkgconfig/Makefile.am:
5881 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5882 * pkgconfig/gst-rtsp-server.pc.in:
5883 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5884 * pkgconfig/gstreamer-rtsp-server.pc.in:
5885 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5886 For consistency with all other modules.
5888 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5890 * gst/rtsp-server/rtsp-client.c:
5891 rtsp-client: update for new map API
5893 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5896 * bindings/Makefile.am:
5897 * bindings/python/Makefile.am:
5898 * bindings/python/arg-types.py:
5899 * bindings/python/codegen/Makefile.am:
5900 * bindings/python/codegen/__init__.py:
5901 * bindings/python/codegen/argtypes.py:
5902 * bindings/python/codegen/code-coverage.py:
5903 * bindings/python/codegen/codegen.py:
5904 * bindings/python/codegen/definitions.py:
5905 * bindings/python/codegen/defsparser.py:
5906 * bindings/python/codegen/docextract.py:
5907 * bindings/python/codegen/docgen.py:
5908 * bindings/python/codegen/fileprefix.override:
5909 * bindings/python/codegen/fileprefixmodule.c:
5910 * bindings/python/codegen/h2def.py:
5911 * bindings/python/codegen/mergedefs.py:
5912 * bindings/python/codegen/mkskel.py:
5913 * bindings/python/codegen/override.py:
5914 * bindings/python/codegen/reversewrapper.py:
5915 * bindings/python/codegen/scmexpr.py:
5916 * bindings/python/rtspserver-types.defs:
5917 * bindings/python/rtspserver.defs:
5918 * bindings/python/rtspserver.override:
5919 * bindings/python/rtspservermodule.c:
5920 * bindings/python/test.py:
5922 python: remove pygst-based python bindings
5923 pygi is the future, apparently.
5925 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
5928 Automatic update of common submodule
5929 From c463bc0 to 7fda524
5931 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5934 Automatic update of common submodule
5935 From 2a59016 to c463bc0
5937 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5940 Automatic update of common submodule
5941 From 0807187 to 2a59016
5943 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5946 Automatic update of common submodule
5947 From 11f0cd5 to 0807187
5949 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5951 * examples/test-auth.c:
5952 example: update for new caps
5954 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5956 * examples/test-video.c:
5957 * gst/rtsp-server/rtsp-client.c:
5958 * gst/rtsp-server/rtsp-media-factory-uri.c:
5959 * gst/rtsp-server/rtsp-media.c:
5960 * gst/rtsp-server/rtsp-media.h:
5961 * gst/rtsp-server/rtsp-session.c:
5962 * gst/rtsp-server/rtsp-session.h:
5963 rtsp-server: port some more to 0.11
5965 Remove bufferlist stuff
5967 Add queue before appsink now that preroll-queue-len is gone.
5968 Update for request pad changes.
5970 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5972 Merge branch 'master' into 0.11
5974 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5976 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5977 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5978 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5980 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5982 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5983 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5984 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5986 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5988 Merge branch 'master' into 0.11
5990 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5992 * gst/rtsp-server/rtsp-media.c:
5993 * gst/rtsp-server/rtsp-media.h:
5994 media: add a seekable boolean
5995 Maintain the seekable state with a new variable instead of reusing the
5998 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6000 * gst/rtsp-server/rtsp-media.c:
6001 Disallow seek in live media
6003 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6005 Merge branch 'master' into 0.11
6007 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6009 * gst/rtsp-server/rtsp-server.c:
6010 #ifdef statements for windows socket creation were missing
6012 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6015 Automatic update of common submodule
6016 From a39eb83 to 11f0cd5
6018 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6021 Automatic update of common submodule
6022 From 605cd9a to a39eb83
6024 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6026 Merge branch 'master' into 0.11
6028 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6030 * gst/rtsp-server/rtsp-client.c:
6031 client: use method to access property
6033 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6035 * gst/rtsp-server/rtsp-media-factory.c:
6036 * gst/rtsp-server/rtsp-media-factory.h:
6037 media-factory: add protocols property
6038 Add a property to configure the allowed protocols in the media created from the
6041 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6043 * gst/rtsp-server/rtsp-media-factory.c:
6044 * gst/rtsp-server/rtsp-media-factory.h:
6045 media-factory: add media-configure signal
6046 Add signal to allow the application to configure the media after it was created
6049 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6051 * gst/rtsp-server/rtsp-client.c:
6052 client: use method to access property
6054 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6056 * gst/rtsp-server/rtsp-media-factory.c:
6057 * gst/rtsp-server/rtsp-media-factory.h:
6058 media-factory: add protocols property
6059 Add a property to configure the allowed protocols in the media created from the
6062 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6064 * gst/rtsp-server/rtsp-media-factory.c:
6065 * gst/rtsp-server/rtsp-media-factory.h:
6066 media-factory: add media-configure signal
6067 Add signal to allow the application to configure the media after it was created
6070 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6072 Merge branch 'master' into 0.11
6074 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6076 * gst/rtsp-server/rtsp-client.c:
6077 client: use media multicast group
6079 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6081 * gst/rtsp-server/rtsp-media-factory.h:
6082 * gst/rtsp-server/rtsp-server.h:
6083 * gst/rtsp-server/rtsp-session-pool.h:
6084 * gst/rtsp-server/rtsp-session.h:
6087 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6089 * gst/rtsp-server/rtsp-client.c:
6090 * gst/rtsp-server/rtsp-sdp.h:
6091 sdp: copy and free the server ip address
6092 Copy and free the server ip address to make memory management easier later.
6094 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6096 * gst/rtsp-server/rtsp-media-factory.c:
6097 media-factory: configure multicast in media
6099 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6101 * gst/rtsp-server/rtsp-media.c:
6102 * gst/rtsp-server/rtsp-media.h:
6103 media: add property for multicast group
6104 Add a property to configure the multicast group in the media.
6105 Based on patches from Marc Leeman and Robert Krakora.
6107 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6109 * gst/rtsp-server/rtsp-media-factory.c:
6110 * gst/rtsp-server/rtsp-media-factory.h:
6111 media-factory: add property for multicast group
6112 Add a property to configure the multicast group in the media factory.
6113 Based on patches from Marc Leeman and Robert Krakora.
6115 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6117 * gst/rtsp-server/rtsp-client.c:
6118 client: do configuration of transport in one place
6119 Move the configuration of the transport destination address to where we also
6120 configure the other bits.
6122 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6124 * gst/rtsp-server/rtsp-client.c:
6125 client: use media multicast group
6127 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6129 * gst/rtsp-server/rtsp-media-factory.h:
6130 * gst/rtsp-server/rtsp-server.h:
6131 * gst/rtsp-server/rtsp-session-pool.h:
6132 * gst/rtsp-server/rtsp-session.h:
6135 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6137 * gst/rtsp-server/rtsp-client.c:
6138 * gst/rtsp-server/rtsp-sdp.h:
6139 sdp: copy and free the server ip address
6140 Copy and free the server ip address to make memory management easier later.
6142 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6144 * gst/rtsp-server/rtsp-media-factory.c:
6145 media-factory: configure multicast in media
6147 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6149 * gst/rtsp-server/rtsp-media.c:
6150 * gst/rtsp-server/rtsp-media.h:
6151 media: add property for multicast group
6152 Add a property to configure the multicast group in the media.
6153 Based on patches from Marc Leeman and Robert Krakora.
6155 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6157 * gst/rtsp-server/rtsp-media-factory.c:
6158 * gst/rtsp-server/rtsp-media-factory.h:
6159 media-factory: add property for multicast group
6160 Add a property to configure the multicast group in the media factory.
6161 Based on patches from Marc Leeman and Robert Krakora.
6163 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6165 * gst/rtsp-server/rtsp-client.c:
6166 client: do configuration of transport in one place
6167 Move the configuration of the transport destination address to where we also
6168 configure the other bits.
6170 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6172 Merge branch 'master' into 0.11
6174 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6176 * gst/rtsp-server/rtsp-client.c:
6177 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6178 The problem occurs when the client abruptly closes the connection without
6179 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6180 server is where the pipeline gets torn down. Since this handler is not called,
6181 the pipeline remains and is up and running. Subsequent clients get their own
6182 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6183 remain up and running. This is a resource leak.
6185 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6187 Merge branch 'master' into 0.11
6189 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6191 * gst/rtsp-server/rtsp-media-factory.c:
6192 * gst/rtsp-server/rtsp-media-factory.h:
6193 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6194 For example, it can be used to retrieve source elements like appsrc, in a more
6195 convenient way than subclassing get_element.
6197 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6199 Merge branch 'master' into 0.11
6201 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6203 * gst/rtsp-server/rtsp-server.c:
6204 rtsp-server: hold on to reference while using object
6206 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6208 * gst/rtsp-server/rtsp-media.c:
6211 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6214 configure: use unstable api
6216 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6218 * gst/rtsp-server/rtsp-client.c:
6219 client: fix reference counting
6221 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6223 * gst/rtsp-server/rtsp-client.c:
6224 * gst/rtsp-server/rtsp-media.c:
6225 fix compiler warnings about unused variables
6227 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6229 * examples/test-launch.c:
6230 * examples/test-readme.c:
6231 * examples/test-uri.c:
6232 * examples/test-video.c:
6233 examples: tell rtsp uri when ready
6235 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6238 Automatic update of common submodule
6239 From 69b981f to 605cd9a
6241 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6243 * gst/rtsp-server/rtsp-client.c:
6244 client: update for buffer API change
6246 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6248 * gst/rtsp-server/Makefile.am:
6249 Makefile.am: 0.10 => @GST_MAJORMINOR@
6251 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6253 * gst/rtsp-server/rtsp-media-factory-uri.c:
6254 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6256 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6258 * gst/rtsp-server/.gitignore:
6259 .gitignore: 0.10 => 0.11
6261 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6263 * gst/rtsp-server/Makefile.am:
6264 Makefile.am: 0.10 => @GST_MAJORMINOR@
6266 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6268 Merge branch 'master' into 0.11
6270 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6273 Automatic update of common submodule
6274 From 9e5bbd5 to 69b981f
6276 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6279 Automatic update of common submodule
6280 From fd35073 to 9e5bbd5
6282 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6285 Automatic update of common submodule
6286 From 46dfcea to fd35073
6288 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-media-factory-uri.c:
6291 * gst/rtsp-server/rtsp-media.c:
6292 media: port to new caps API
6294 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6296 Merge branch 'master' into 0.11
6298 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6300 * bindings/vala/gst-rtsp-server-0.10.vapi:
6301 Updated Vala bindings.
6302 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6304 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6306 * gst/rtsp-server/rtsp-server.c:
6307 * gst/rtsp-server/rtsp-server.h:
6308 Add a signal for newly connected clients.
6309 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6311 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6313 * bindings/python/rtspserver.override:
6314 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6316 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6318 * gst/rtsp-server/Makefile.am:
6319 * gst/rtsp-server/rtsp-client.c:
6320 * gst/rtsp-server/rtsp-funnel.c:
6321 * gst/rtsp-server/rtsp-funnel.h:
6322 * gst/rtsp-server/rtsp-media.c:
6323 rtsp-server: port to 0.11
6325 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6330 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6332 Merge branch 'master' into 0.11
6337 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6340 Automatic update of common submodule
6341 From c3cafe1 to 46dfcea
6343 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6345 * bindings/python/Makefile.am:
6346 * bindings/python/rtspserver.defs:
6347 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6349 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6351 * bindings/python/arg-types.py:
6352 python bindings: add GstRTSPUrlParam
6353 Needed to implement MediaFactory virtual proxies
6355 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6357 * bindings/python/arg-types.py:
6358 python bindings: fix returning GstRTSPUrl types
6360 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6362 * bindings/python/arg-types.py:
6363 python bindings: add arg type for GstRTSPUrl
6365 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6367 * bindings/python/rtspserver.defs:
6368 python bindings: fix the definition of MediaFactory.collect_stream
6370 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6373 Automatic update of common submodule
6374 From 1ccbe09 to c3cafe1
6376 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6379 Automatic update of common submodule
6380 From 193b717 to 1ccbe09
6382 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6385 Automatic update of common submodule
6386 From b77e2bf to 193b717
6388 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6391 build: Include lcov.mak to allow test coverage report generation
6393 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6396 Automatic update of common submodule
6397 From d8814b6 to b77e2bf
6399 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6402 Automatic update of common submodule
6403 From 6aaa286 to d8814b6
6405 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6408 Automatic update of common submodule
6409 From 6aec6b9 to 6aaa286
6411 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6414 autogen: wingo signed comment
6416 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6418 * gst/rtsp-server/rtsp-session-pool.c:
6419 session: use full charset for RTSP session ID
6420 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6421 session ID more difficult.
6422 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6424 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6426 * gst/rtsp-server/Makefile.am:
6427 rtsp-server: Don't install the funnel header
6429 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6432 Automatic update of common submodule
6433 From 1de7f6a to 6aec6b9
6435 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6438 configure: require core/base 0.10.31
6439 Needed at least for gst_plugin_feature_rank_compare_func().
6441 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6444 Automatic update of common submodule
6445 From f94d739 to 1de7f6a
6447 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6449 * gst/rtsp-server/rtsp-media.c:
6450 media: remove more unused code
6452 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6454 * gst/rtsp-server/rtsp-media.c:
6455 * gst/rtsp-server/rtsp-media.h:
6456 media: remove duplicate filtering
6457 Remove the duplicate filtering code now that we have a released -good version.
6458 Give a warning instead.
6460 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6462 * gst/rtsp-server/rtsp-media-factory.c:
6463 * gst/rtsp-server/rtsp-media.c:
6464 media: fix default buffer size
6466 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6468 * gst/rtsp-server/rtsp-media-factory.c:
6469 * gst/rtsp-server/rtsp-media-factory.h:
6470 media-factory: add property to configure the buffer-size
6471 Add a property to configure the kernel UDP buffer size.
6473 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6475 * gst/rtsp-server/rtsp-media.c:
6476 * gst/rtsp-server/rtsp-media.h:
6477 media: add property to configure kernel buffer sizes
6478 Add a property to configure the kernel UDP buffer size.
6480 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6483 configure: set PYGOBJECT_REQ before using it
6484 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6486 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6489 docs: recursive into sub-directories on 'make upload'
6491 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6493 * docs/libs/gst-rtsp-server-docs.sgml:
6494 * docs/version.entities.in:
6495 docs: mention full version these docs are for, not just major-minor
6497 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6502 === release 0.10.8 ===
6504 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6509 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6511 * gst/rtsp-server/rtsp-server.c:
6512 rtsp-server: clarify docs a little
6514 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6516 * gst/rtsp-server/rtsp-media.c:
6517 media: init debug category before starting thread
6519 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6521 * gst/rtsp-server/rtsp-auth.c:
6522 auth: add realm to make it more spec compliant
6524 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6526 * gst/rtsp-server/rtsp-server.c:
6527 * gst/rtsp-server/rtsp-server.h:
6530 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6532 * examples/test-video.c:
6533 example: improve example docs a little
6535 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6537 * gst/rtsp-server/rtsp-server.c:
6538 server: ensure the watch has a ref to the server
6540 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6542 * gst/rtsp-server/rtsp-server.c:
6543 server: simpify channel function
6545 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6547 * gst/rtsp-server/rtsp-server.c:
6548 * gst/rtsp-server/rtsp-server.h:
6549 server: simplify management of channel and source
6550 We don't need to keep around the channel and source objects. Let the mainloop
6551 and the source manage the source and channel respectively.
6553 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6559 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6562 * tests/Makefile.am:
6563 * tests/test-cleanup.c:
6564 tests: add tests directory and cleanup test
6566 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6568 * gst/rtsp-server/rtsp-media-factory-uri.c:
6569 * gst/rtsp-server/rtsp-media-factory.c:
6570 * gst/rtsp-server/rtsp-media-mapping.c:
6571 * gst/rtsp-server/rtsp-media.c:
6572 * gst/rtsp-server/rtsp-session-pool.c:
6573 * gst/rtsp-server/rtsp-session.c:
6574 server: improve debugging in various objects
6576 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6578 * gst/rtsp-server/rtsp-server.c:
6579 server: chain up to the parent finalize
6581 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6583 * bindings/python/rtspserver-types.defs:
6584 * bindings/python/rtspserver.defs:
6585 * bindings/python/rtspserver.override:
6586 * bindings/python/test.py:
6587 gst-rtsp-server: update python bindings
6589 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6591 * gst/rtsp-server/rtsp-client.c:
6592 client: use the response from the clientstate
6593 Create the response object only once and store in the client state.
6594 Make all methods use the state response,
6596 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6598 * gst/rtsp-server/rtsp-server.c:
6599 server: use signal to keep track of clients
6600 Keep track of all the clients that the server creates and remove them when they
6601 fire the 'closed' signal.
6603 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6605 * gst/rtsp-server/rtsp-client.c:
6606 * gst/rtsp-server/rtsp-client.h:
6607 client: emit signal when closing
6609 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6611 * examples/.gitignore:
6612 * examples/Makefile.am:
6613 * examples/test-auth.c:
6614 * examples/test-video.c:
6615 * gst/rtsp-server/rtsp-auth.c:
6616 * gst/rtsp-server/rtsp-auth.h:
6617 * gst/rtsp-server/rtsp-client.c:
6618 * gst/rtsp-server/rtsp-media-factory.c:
6619 * gst/rtsp-server/rtsp-media.c:
6620 * gst/rtsp-server/rtsp-media.h:
6621 * gst/rtsp-server/rtsp-session-pool.h:
6622 * gst/rtsp-server/rtsp-session.h:
6623 media: enable per factory authorisations
6624 Allow for adding a GstRTSPAuth on the factory and media level and check
6625 permissions when accessing the factory.
6626 Add hints to the auth methods for future more fine grained authorisation.
6627 Add example application for per factory authentication.
6629 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6631 * gst/rtsp-server/rtsp-auth.c:
6632 * gst/rtsp-server/rtsp-auth.h:
6633 * gst/rtsp-server/rtsp-client.c:
6634 * gst/rtsp-server/rtsp-client.h:
6635 * gst/rtsp-server/rtsp-params.c:
6636 * gst/rtsp-server/rtsp-params.h:
6637 rtsp-server: Pass ClientState structure arround
6638 Pass the collected information for the ongoing request in a GstRTSPClientState
6639 structure that we can then pass around to simplify the method arguments. This
6640 will also be handy when we implement logging functionality.
6642 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6644 * gst/rtsp-server/rtsp-media-factory.c:
6645 * gst/rtsp-server/rtsp-media-factory.h:
6646 media-factory: add methods to configure authorisation
6648 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-client.c:
6651 client: unref auth in finalize
6653 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6655 * gst/rtsp-server/rtsp-server.c:
6656 server: unref auth in finalize
6658 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6660 * docs/libs/gst-rtsp-server-docs.sgml:
6661 * docs/libs/gst-rtsp-server-sections.txt:
6662 * docs/libs/gst-rtsp-server.types:
6665 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6667 * gst/rtsp-server/rtsp-server.c:
6668 * gst/rtsp-server/rtsp-server.h:
6669 server: separate create and accept
6670 Create separate create and accept methods so that subclasses can create custom
6672 Configure the server in the client object and prepare for keeping track of
6675 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6677 * gst/rtsp-server/rtsp-client.c:
6678 * gst/rtsp-server/rtsp-client.h:
6679 client: add support for setting the server.
6680 Add support for keeping a ref to the server that started this client
6683 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6685 * gst/rtsp-server/rtsp-auth.c:
6686 auth: fix memleak and add some docs
6687 Fix a memleak of the basic auth token.
6688 Add docs for the helper function
6690 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6692 * gst/rtsp-server/rtsp-auth.c:
6693 * gst/rtsp-server/rtsp-auth.h:
6694 * gst/rtsp-server/rtsp-client.c:
6695 client: delegate setup of auth to the manager
6696 Delegate the configuration of the authentication tokens to the manager object
6699 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6701 * examples/test-video.c:
6702 * gst/rtsp-server/Makefile.am:
6703 * gst/rtsp-server/rtsp-auth.c:
6704 * gst/rtsp-server/rtsp-auth.h:
6705 * gst/rtsp-server/rtsp-client.c:
6706 * gst/rtsp-server/rtsp-client.h:
6707 * gst/rtsp-server/rtsp-server.c:
6708 * gst/rtsp-server/rtsp-server.h:
6709 auth: add authentication object
6710 Add an object that can check the authorization of requests.
6711 Implement basic authentication.
6712 Add example authentication to test-video
6714 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6716 * gst/rtsp-server/rtsp-server.c:
6717 * gst/rtsp-server/rtsp-server.h:
6718 server: move includes back
6719 the includes are needed for sockaddr_in.
6721 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6723 * gst/rtsp-server/rtsp-client.c:
6724 * gst/rtsp-server/rtsp-client.h:
6725 * gst/rtsp-server/rtsp-server.c:
6726 * gst/rtsp-server/rtsp-server.h:
6727 rtsp: move network includes where they are needed
6729 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6731 * gst/rtsp-server/rtsp-media.h:
6732 rtsp-media.h: Minor corrections in comments.
6735 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6738 Automatic update of common submodule
6739 From e572c87 to f94d739
6741 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6745 * docs/libs/.gitignore:
6746 * examples/.gitignore:
6747 * gst/rtsp-server/.gitignore:
6750 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6752 * docs/libs/Makefile.am:
6753 docs: We don't build ps/pdf for API reference docs
6755 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6758 Automatic update of common submodule
6759 From ccbaa85 to e572c87
6761 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6764 Automatic update of common submodule
6765 From 46445ad to ccbaa85
6767 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6769 * gst/rtsp-server/Makefile.am:
6770 * gst/rtsp-server/fs-funnel.c:
6771 * gst/rtsp-server/fs-funnel.h:
6772 * gst/rtsp-server/rtsp-funnel.c:
6773 * gst/rtsp-server/rtsp-funnel.h:
6774 * gst/rtsp-server/rtsp-media.c:
6775 funnel: rename fsfunnel to rtspfunnel
6776 Rename the funnel to avoid conflicts with the farsight one.
6778 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6780 * gst/rtsp-server/Makefile.am:
6781 * gst/rtsp-server/fs-funnel.c:
6782 * gst/rtsp-server/fs-funnel.h:
6783 * gst/rtsp-server/rtsp-media.c:
6784 rtsp-media: add and use fsfunnel
6785 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6786 select-all property that we need.
6788 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6790 * gst/rtsp-server/Makefile.am:
6791 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6792 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6793 for the g-ir-compiler, rather than just assuming the env var has
6796 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6803 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6805 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6808 * gst/rtsp-server/Makefile.am:
6809 gobject-introspection: fix g-i build for uninstalled setup
6810 Requires gst-plugins-base git (> 0.10.31.2).
6812 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6814 * examples/test-uri.c:
6815 examples: add some more options and comments
6817 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6819 * gst/rtsp-server/rtsp-media-factory-uri.c:
6820 factory-uri: use right property type
6822 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6824 * gst/rtsp-server/rtsp-media-factory-uri.c:
6825 factory-uri: attempt to configure buffer-lists
6826 Attempt to configure buffer lists in the payloader for improved performance.
6828 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6830 * gst/rtsp-server/rtsp-media.c:
6831 media: attempt to configure bigger UDP buffers
6832 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6833 send buffers with high bitrate streams.
6835 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6837 * gst/rtsp-server/rtsp-client.c:
6838 client: use the socket length from getsockname
6839 Use the length returned by getsockname to perform the getnameinfo call because
6840 the size can depend on the socket type and platform.
6843 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6845 * docs/libs/gst-rtsp-server-docs.sgml:
6846 * docs/libs/gst-rtsp-server-sections.txt:
6847 docs: add uri factory to the docs
6849 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6851 * gst/rtsp-server/rtsp-client.c:
6852 * gst/rtsp-server/rtsp-media.h:
6855 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6857 * gst/rtsp-server/rtsp-client.c:
6858 * gst/rtsp-server/rtsp-media.c:
6859 * gst/rtsp-server/rtsp-media.h:
6860 * gst/rtsp-server/rtsp-session.c:
6861 * gst/rtsp-server/rtsp-session.h:
6862 rtsp-server: add support for buffer lists
6863 Add support for sending bufferlists received from appsink.
6866 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6868 * gst/rtsp-server/rtsp-client.c:
6869 * gst/rtsp-server/rtsp-media.c:
6870 * gst/rtsp-server/rtsp-media.h:
6871 * gst/rtsp-server/rtsp-sdp.c:
6872 media: make method to retrieve the play range
6873 Make a method to retrieve the playback range so that we can conditionally create
6874 a different range for the SDP and the PLAY requests.
6876 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6878 * gst/rtsp-server/rtsp-media.c:
6879 * gst/rtsp-server/rtsp-media.h:
6880 media: add signal to notify of state changes
6882 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6884 * gst/rtsp-server/rtsp-client.h:
6885 client: cleanup headers
6887 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6889 * gst/rtsp-server/rtsp-client.c:
6892 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6894 * gst/rtsp-server/rtsp-media-factory-uri.c:
6895 * gst/rtsp-server/rtsp-media-factory-uri.h:
6896 factory-uri: add support for gstpay
6897 Add an option to prefer gstpay over decoder + raw payloader.
6899 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6901 * gst/rtsp-server/rtsp-media-factory-uri.c:
6902 * gst/rtsp-server/rtsp-media-factory-uri.h:
6903 factory-uri: rework the autoplugger.
6904 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
6907 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6909 * gst/rtsp-server/rtsp-media-factory-uri.c:
6910 factory-uri: use better factory filter
6911 Make better payloader filter based on autoplug rank and RTP use case.
6913 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6916 Automatic update of common submodule
6917 From 169462a to 46445ad
6919 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6921 * gst/rtsp-server/rtsp-server.c:
6922 server: set SO_REUSEADDR before bind
6923 Set the SO_REUSEADDR _before_ bind() to make it actually work.
6925 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6927 * gst/rtsp-server/rtsp-media.c:
6928 * gst/rtsp-server/rtsp-media.h:
6929 media: emit prepared signal when prepared
6930 Make a 'prepared' signal and emit it when we successfully prepared the element.
6931 This signal can be used to configure the media object after it has been prepared
6934 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
6937 Automatic update of common submodule
6938 From 011bcc8 to 169462a
6940 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
6942 python an optional dependency
6943 * configure.ac: Move up valgrind and g-i checks. Make the python
6944 dependency optional, as it was before.
6946 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6948 Merge branch 'master' into 0.11
6953 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6955 * gst/rtsp-server/rtsp-media.c:
6956 media: update range when active clients changed
6957 When we changed the number of active clients, update the current range
6958 information because we want the second client connecting to a shared resource
6959 continue from where the stream currently.
6961 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6963 * gst/rtsp-server/rtsp-media-factory-uri.c:
6964 * gst/rtsp-server/rtsp-media-factory-uri.h:
6965 factory-uri: add colorspace and fix pt
6966 Rework the way we pass data to the autoplugger.
6967 When we have raw caps, plug a converter element to make pluggin to raw
6968 payloaders more successful.
6969 Make sure all dynamically plugged payloaders have a unique payload types.
6971 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6973 * examples/Makefile.am:
6974 * examples/test-uri.c:
6975 example: add example of the uri factory
6977 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6979 * gst/rtsp-server/Makefile.am:
6980 * gst/rtsp-server/rtsp-media-factory-uri.c:
6981 * gst/rtsp-server/rtsp-media-factory-uri.h:
6982 * gst/rtsp-server/rtsp-server.h:
6983 factory-uri: add a factory to stream any URI
6984 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
6987 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6989 * gst/rtsp-server/rtsp-media.c:
6990 * gst/rtsp-server/rtsp-media.h:
6991 media: ignore spurious ASYNC_DONE messages
6992 When we are dynamically adding pads, the addition of the udpsrc elements will
6993 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
6994 the real ASYNC_DONE when everything is prerolled.
6996 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6998 * gst/rtsp-server/rtsp-media-factory.c:
6999 * gst/rtsp-server/rtsp-media-factory.h:
7000 media-factory: make lock macro
7002 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7004 * gst/rtsp-server/rtsp-client.c:
7005 rtsp-server: Remove unused variable and dead assignment
7007 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7009 * examples/test-launch.c:
7010 * examples/test-mp4.c:
7011 * examples/test-ogg.c:
7012 * examples/test-readme.c:
7013 * examples/test-sdp.c:
7014 * examples/test-video.c:
7015 examples: Run gst-indent
7017 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7019 * gst/rtsp-server/rtsp-client.c:
7020 * gst/rtsp-server/rtsp-media-factory.c:
7021 * gst/rtsp-server/rtsp-media-mapping.c:
7022 * gst/rtsp-server/rtsp-media.c:
7023 * gst/rtsp-server/rtsp-params.c:
7024 * gst/rtsp-server/rtsp-sdp.c:
7025 * gst/rtsp-server/rtsp-server.c:
7026 * gst/rtsp-server/rtsp-session-pool.c:
7027 * gst/rtsp-server/rtsp-session.c:
7028 rtsp-server: Run gst-indent
7029 Since it wasn't using the upstream common previously, there was no
7030 indentation check before commiting.
7032 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7034 * gst/rtsp-server/rtsp-media-mapping.h:
7035 * gst/rtsp-server/rtsp-media.c:
7036 * gst/rtsp-server/rtsp-media.h:
7037 * gst/rtsp-server/rtsp-sdp.c:
7038 * gst/rtsp-server/rtsp-session-pool.h:
7039 * gst/rtsp-server/rtsp-session.c:
7040 * gst/rtsp-server/rtsp-session.h:
7041 rtsp-server: Some more doc fixups
7043 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7046 Makefile: Add cruft-cleaning support
7048 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7053 * docs/libs/Makefile.am:
7054 * docs/libs/gst-rtsp-server-docs.sgml:
7055 * docs/libs/gst-rtsp-server-sections.txt:
7056 * docs/libs/gst-rtsp-server.types:
7057 * docs/version.entities.in:
7058 docs: Add gtk-doc build system
7060 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7062 * gst/rtsp-server/Makefile.am:
7063 Makefile.am: Use standard GIR make behaviour
7065 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7069 autogen/configure: Bring more in sync to standard gst module behaviour
7071 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7073 * gst/rtsp-server/rtsp-media.c:
7074 media: warn and fail when gstrtpbin is not found
7076 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7079 configure: open 0.11 branch
7081 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7085 Add common submodule
7087 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7090 * common/Makefile.am:
7091 * common/c-to-xml.py:
7093 * common/coverage/coverage-report-entry.pl:
7094 * common/coverage/coverage-report.pl:
7095 * common/coverage/coverage-report.xsl:
7096 * common/coverage/lcov.mak:
7097 * common/gettext.patch:
7098 * common/glib-gen.mak:
7099 * common/gst-autogen.sh:
7100 * common/gst-xmlinspect.py:
7102 * common/gstdoc-scangobj:
7103 * common/gtk-doc-plugins.mak:
7104 * common/gtk-doc.mak:
7105 * common/m4/.gitignore:
7106 * common/m4/Makefile.am:
7108 * common/m4/as-ac-expand.m4:
7109 * common/m4/as-auto-alt.m4:
7110 * common/m4/as-compiler-flag.m4:
7111 * common/m4/as-compiler.m4:
7112 * common/m4/as-docbook.m4:
7113 * common/m4/as-libtool-tags.m4:
7114 * common/m4/as-libtool.m4:
7115 * common/m4/as-python.m4:
7116 * common/m4/as-scrub-include.m4:
7117 * common/m4/as-version.m4:
7118 * common/m4/ax_create_stdint_h.m4:
7119 * common/m4/check.m4:
7120 * common/m4/glib-gettext.m4:
7121 * common/m4/gst-arch.m4:
7122 * common/m4/gst-args.m4:
7123 * common/m4/gst-check.m4:
7124 * common/m4/gst-debuginfo.m4:
7125 * common/m4/gst-default.m4:
7126 * common/m4/gst-doc.m4:
7127 * common/m4/gst-error.m4:
7128 * common/m4/gst-feature.m4:
7129 * common/m4/gst-function.m4:
7130 * common/m4/gst-gettext.m4:
7131 * common/m4/gst-glib2.m4:
7132 * common/m4/gst-libxml2.m4:
7133 * common/m4/gst-plugindir.m4:
7134 * common/m4/gst-valgrind.m4:
7135 * common/m4/gtk-doc.m4:
7136 * common/m4/introspection.m4:
7138 * common/mangle-tmpl.py:
7139 * common/plugins.xsl:
7141 * common/release.mak:
7142 * common/scangobj-merge.py:
7143 * common/upload.mak:
7144 common: Remove static version
7146 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7148 * common/m4/introspection.m4:
7149 Update introspection.m4 to match usage
7151 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7155 Remove old stuff from the README
7157 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7162 === release 0.10.7 ===
7164 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7169 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7171 * examples/test-ogg.c:
7172 test-ogg: remove parsers
7173 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7174 buffers with timestamps. Using the parsers also seems to break things.
7176 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7178 * bindings/vala/gst-rtsp-server-0.10.vapi:
7179 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7180 Updated Vala bindings
7182 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7184 * common/m4/introspection.m4:
7186 * gst/rtsp-server/Makefile.am:
7187 Added initial gobject-introspection support
7189 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7191 * gst/rtsp-server/rtsp-media-factory.c:
7192 media-factory: don't use host for shared hash key
7193 When we generate the key to share made between connections, don't include the
7194 host used to connect so that we can share media even if between clients that
7195 connected with localhost and ones with the ip address.
7197 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7199 * bindings/vala/Makefile.am:
7200 build: fix distcheck
7202 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7204 * bindings/vala/gst-rtsp-server-0.10.vapi:
7205 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7206 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7207 Update Vala bindings
7209 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7211 * bindings/vala/Makefile.am:
7213 Fix configure checks and installation location for Vala bindings
7216 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7221 === release 0.10.6 ===
7223 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7226 configure: release 0.10.6
7228 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7230 * gst/rtsp-server/rtsp-media.c:
7231 media: help the compiler a little
7233 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7235 * gst/rtsp-server/rtsp-media.c:
7236 * gst/rtsp-server/rtsp-media.h:
7237 * gst/rtsp-server/rtsp-session.c:
7238 media: cleanup media transport before freeing
7239 Cleanup the media transport data before freeing. In particular, remove the qdata
7240 from the rtpsource object.
7242 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7244 * gst/rtsp-server/rtsp-media-factory.c:
7245 * gst/rtsp-server/rtsp-media-factory.h:
7246 * gst/rtsp-server/rtsp-media.c:
7247 * gst/rtsp-server/rtsp-media.h:
7248 media-factory: add eos-shutdown property
7249 Add an eos-shutdown property that will send an EOS to the pipeline before
7250 shutting it down. This allows for nice cleanup in case of a muxer.
7253 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7255 * gst/rtsp-server/rtsp-media.c:
7256 * gst/rtsp-server/rtsp-media.h:
7257 media: use multiudpsink send-duplicates when we can
7258 If we have a new enough multiudpsink with the send-duplicates property, use this
7259 instead of doing our own filtering. Our custom filtering code should eventually
7260 be removed when we can depend on a released -good.
7262 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7264 * gst/rtsp-server/rtsp-media.c:
7265 media: don't leak destinations
7266 Refactor and cleanup the destinations array when the stream is destroyed.
7268 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7270 * gst/rtsp-server/rtsp-media.c:
7271 * gst/rtsp-server/rtsp-media.h:
7272 media: don't add udp addresses multiple times
7273 Keep track of the udp addresses we added to udpsink and never add the same udp
7274 destination twice. This avoids duplicate packets when using multicast.
7276 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7278 * gst/rtsp-server/rtsp-server.c:
7279 server: disable use of SO_LINGER
7280 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7281 server close()s the connection.
7283 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7285 * gst/rtsp-server/rtsp-server.c:
7286 server: use 5 second linger period in SO_LINGER
7287 Wait 5 seconds before clearing the send buffers and reseting the connection with
7288 the client when we do a close. This should be enough time to get the message to
7292 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7294 * gst/rtsp-server/rtsp-server.c:
7295 server: use SO_LINGER
7296 SO_LINGER on the socket will make sure that any pending data on the socket is
7297 flushed ASAP and that the socket connection is reset. This makes sure that the
7298 socket can be reused immediately.
7301 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7304 README: add blurb about shared media factories
7306 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7308 * gst/rtsp-server/rtsp-media.c:
7309 Add stdlib.h for atoi()
7311 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7313 * bindings/python/Makefile.am:
7314 * bindings/vala/Makefile.am:
7315 build: distcheck fixes
7316 Fix 'make distcheck', somewhat (it still fails because it tries to
7317 install files into /usr/share/vala/vapi/ irrespective of the
7320 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7323 configure: bump core/base requirements to released version
7324 Makes things less confusing for people.
7326 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7329 configure: fail if GStreamer core/base requirements are not met
7331 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7333 * gst/rtsp-server/rtsp-client.c:
7334 client: improve client cleanups
7335 Make sure the session does not timeout when using TCP. We need to do this
7336 because quicktime player does not send RTCP for some reason in tunneled
7338 Refactor some cleanup code.
7341 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * gst/rtsp-server/rtsp-session.c:
7344 * gst/rtsp-server/rtsp-session.h:
7345 session: add support for prevent session timeouts
7346 Add an atomix counter to prevent session timeouts when we are, for example,
7349 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7351 * gst/rtsp-server/rtsp-client.c:
7352 client: fix unlink on session timeouts
7353 When our session times out, make sure we unlink all streams in this
7355 Remove the tunnelid when closing the connection.
7357 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7359 * gst/rtsp-server/rtsp-session.c:
7360 session: small cleanups
7362 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7364 * gst/rtsp-server/rtsp-client.c:
7365 client: handle lost_tunnel callbacks
7366 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7367 hashtable so that we can reuse it for when the client reopens the POST
7369 Close the connection after a TEARDOWN.
7370 Make sure or watchid is cleared when the watch is removed.
7373 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7375 * gst/rtsp-server/rtsp-client.c:
7376 * gst/rtsp-server/rtsp-media.c:
7377 * gst/rtsp-server/rtsp-sdp.c:
7378 rtsp-server: add more support for multicast
7380 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7383 * gst/rtsp-server/rtsp-media.c:
7384 * gst/rtsp-server/rtsp-media.h:
7385 media: allow configuration of allowed lower transport
7387 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7389 * gst/rtsp-server/rtsp-client.h:
7390 * gst/rtsp-server/rtsp-media.c:
7391 * gst/rtsp-server/rtsp-media.h:
7392 * gst/rtsp-server/rtsp-sdp.c:
7393 * gst/rtsp-server/rtsp-sdp.h:
7394 * gst/rtsp-server/rtsp-server.c:
7395 rtsp: keep track of server ip and ipv6
7396 Keep track of how the client connected to the server and setup the udp ports
7397 with the same protocol.
7398 Copy the server ip address in the SDP so that clients can send RTCP back to
7401 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7403 * gst/rtsp-server/rtsp-session.c:
7406 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7408 * gst/rtsp-server/rtsp-client.c:
7409 client: use right size for malloc
7411 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7413 * gst/rtsp-server/rtsp-server.c:
7414 server: comment ipv6 server listening address
7416 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7418 * gst/rtsp-server/rtsp-media.c:
7419 media: allow for ipv6 sockets
7421 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7423 * gst/rtsp-server/rtsp-server.c:
7424 * gst/rtsp-server/rtsp-server.h:
7425 server: rework server part
7426 Allow setting a bind address, make sure we can deal with ipv6.
7427 Remove the port property and change with the service property.
7429 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/rtsp-server/rtsp-media.h:
7432 media: update comments a little
7434 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-client.c:
7437 client: make content-base better
7438 Use the URI formatting functions to make a content-base. Also make sure that
7439 there is a trailing / at the end.
7441 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7443 * gst/rtsp-server/rtsp-client.c:
7444 client: guard against invalid paths
7446 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7448 * examples/test-video.c:
7449 test: catch server bind errors
7451 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7453 * gst/rtsp-server/rtsp-media.c:
7454 rtspmedia: emit "unprepared" if _prepare fails.
7455 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7456 media object is removed from its factory's cache.
7458 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7460 * gst/rtsp-server/rtsp-media.c:
7461 media: collect media position when seek completes
7463 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7465 * gst/rtsp-server/rtsp-client.c:
7466 client: call unlink_streams in client finalize
7469 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7471 * gst/rtsp-server/rtsp-media.c:
7472 media: limit the time to wait to something huge
7473 Avoid waiting forever but limit the timeout to 20 seconds.
7475 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7477 * gst/rtsp-server/rtsp-sdp.c:
7478 sdp: reindent and check for prepared status
7480 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7482 * gst/rtsp-server/rtsp-media.c:
7483 * gst/rtsp-server/rtsp-media.h:
7484 * gst/rtsp-server/rtsp-session.c:
7485 media: avoid doing _get_state() for state changes
7486 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7487 until the media is prerolled or in error. This avoids doing a blocking call of
7488 gst_element_get_state() that can cause lockups when there is an error.
7491 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7493 * gst/rtsp-server/rtsp-media.c:
7496 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7498 * gst/rtsp-server/rtsp-media-factory.c:
7499 media-factory: better error handling
7500 Improve the error handling a bit.
7502 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7504 * gst/rtsp-server/rtsp-client.c:
7505 client: rework transport parsing
7506 Rework the transport parsing code so that we can ignore transports we don't
7507 support instead of just picking the first one we can parse.
7508 Configure a (for now hardcoded) destination for multicast transports.
7510 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7512 * gst/rtsp-server/rtsp-media.c:
7513 media: set multicast sink parameters
7514 Disable loop and automatic multicast join on the udpsink elements.
7515 Add some more debug info.
7516 Reset some state variables in the right place.
7517 Use the right port numbers for multicast.
7519 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7521 * gst/rtsp-server/rtsp-session.c:
7522 session: handle transport setup correctly
7523 Handle UDP, MCAST and TCP transport negotiation more correctly.
7524 Store the server session SSRC in the transport.
7526 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7528 * gst/rtsp-server/rtsp-client.c:
7529 rtsp-client: implement error_full
7530 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7533 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7536 * gst/rtsp-server/rtsp-client.c:
7537 * gst/rtsp-server/rtsp-server.c:
7538 docs: update docs and comments
7540 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7542 * gst/rtsp-server/rtsp-sdp.c:
7543 sdp: make server work better when behind a proxy
7545 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7547 * gst/rtsp-server/rtsp-client.c:
7548 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7550 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7552 * gst/rtsp-server/rtsp-client.c:
7553 * gst/rtsp-server/rtsp-media-factory.c:
7554 * gst/rtsp-server/rtsp-media-mapping.c:
7555 * gst/rtsp-server/rtsp-media.c:
7556 * gst/rtsp-server/rtsp-server.c:
7557 * gst/rtsp-server/rtsp-session-pool.c:
7558 * gst/rtsp-server/rtsp-session.c:
7559 Use GStreamer's debugging subsystem
7561 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7563 * gst/rtsp-server/rtsp-media-factory.c:
7564 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7566 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7571 === release 0.10.5 ===
7573 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7578 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7581 configure: bump required versions
7583 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7585 * gst/rtsp-server/rtsp-client.c:
7586 client: call weak-unref on client->sessions from finalize
7589 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7591 * gst/rtsp-server/rtsp-media.c:
7592 media: Fixed crasher where caps got unref'ed too often
7594 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7597 * pkgconfig/.gitignore:
7598 * pkgconfig/Makefile.am:
7599 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7600 Added pkg-config file to use gst-rtsp-server uninstalled
7602 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7604 * gst/rtsp-server/rtsp-media.c:
7605 media: add some docs
7607 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7609 * gst/rtsp-server/rtsp-client.c:
7610 rtsp: Use gst_rtsp_watch_send_message().
7611 Use gst_rtsp_watch_send_message() since the old API which used
7612 gst_rtsp_watch_queue_message() has been deprecated.
7614 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7619 === release 0.10.4 ===
7621 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7626 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7628 * gst/rtsp-server/rtsp-client.c:
7629 * gst/rtsp-server/rtsp-session.c:
7630 * gst/rtsp-server/rtsp-session.h:
7631 rtsp: allocate channels in TCP mode
7632 When the client does not provide us with channels in TCP mode, allocate channels
7635 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7637 * gst/rtsp-server/rtsp-client.c:
7638 client: don't crash when tunnelid is missing
7639 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7640 don't crash but return an error response to the client.
7643 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7645 * bindings/vala/gst-rtsp-server-0.10.vapi:
7646 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7647 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7648 bindings: update vala bindings with new method
7650 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7652 * gst/rtsp-server/rtsp-session-pool.c:
7653 * gst/rtsp-server/rtsp-session-pool.h:
7654 sessionpool: add function to filter sessions
7655 Add generic function to retrieve/remove sessions.
7657 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7660 configure: bump core/base requirements to release
7662 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7664 * gst/rtsp-server/rtsp-media.c:
7665 media: fix indentation
7667 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7669 * gst/rtsp-server/rtsp-media.c:
7670 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7672 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7674 * gst/rtsp-server/rtsp-media.c:
7675 set state and remove elements of media in for loop
7677 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7679 * bindings/vala/gst-rtsp-server-0.10.vapi:
7680 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7681 Added gst_rtsp_media_remove_elements function to Vala bindings
7683 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7685 * gst/rtsp-server/rtsp-media.c:
7686 * gst/rtsp-server/rtsp-media.h:
7687 Added gst_rtsp_media_remove_elements function
7689 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7691 * gst/rtsp-server/rtsp-media.c:
7692 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7694 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7696 * bindings/vala/gst-rtsp-server-0.10.vapi:
7697 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7698 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7699 Updated Vala bindings
7701 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7703 * gst/rtsp-server/rtsp-media.c:
7704 * gst/rtsp-server/rtsp-media.h:
7705 Added vmethod unprepare to GstRTSPMedia
7706 The default implementation sets the state of the pipeline to GST_STATE_NULL
7708 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7710 * gst/rtsp-server/rtsp-media-factory.c:
7711 * gst/rtsp-server/rtsp-media-factory.h:
7712 Made collect_streams function public
7714 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7716 * gst/rtsp-server/rtsp-media-factory.c:
7717 * gst/rtsp-server/rtsp-media-factory.h:
7718 * gst/rtsp-server/rtsp-media.c:
7719 Added vmethod create_pipeline to GstRTSPMediaFactory
7720 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7722 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7724 * gst/rtsp-server/rtsp-client.c:
7725 client: use g_source_destroy()
7726 We need to use g_source_destroy() because we might have added the source to a
7727 different main context than the default one.
7729 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7731 * gst/rtsp-server/Makefile.am:
7732 * gst/rtsp-server/rtsp-client.c:
7733 * gst/rtsp-server/rtsp-params.c:
7734 * gst/rtsp-server/rtsp-params.h:
7735 rtsp: prepare for handling GET/SET_PARAMETER
7736 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7738 Fix return codes of handlers.
7740 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7742 * gst/rtsp-server/rtsp-media.c:
7743 media: don't leak session pads
7745 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7747 * gst/rtsp-server/rtsp-media.c:
7748 media: clean up the messages a bit
7750 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7752 * gst/rtsp-server/rtsp-sdp.c:
7753 sdp: warn and skip streams without media
7755 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7757 * bindings/vala/gst-rtsp-server-0.10.vapi:
7758 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7759 vala: Fixed typo in header file of RTSPMediaStream
7761 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7763 * gst/rtsp-server/rtsp-media.c:
7766 Make dumping RTCP stats configurable
7768 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7770 * gst/rtsp-server/rtsp-media.c:
7771 media: be less verbose and leak less
7773 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7775 * gst/rtsp-server/rtsp-media.c:
7776 media: don't leak the destination address
7778 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7780 * gst/rtsp-server/rtsp-client.c:
7781 * gst/rtsp-server/rtsp-media.c:
7782 * gst/rtsp-server/rtsp-media.h:
7783 * gst/rtsp-server/rtsp-session.c:
7784 * gst/rtsp-server/rtsp-session.h:
7785 rtsp: use RTCP to keep the session alive
7786 Use the RTCP rtcp-from stats field to find the associated session and use this
7787 to keep the session alive.
7789 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7791 * gst/rtsp-server/rtsp-session.c:
7792 session: add 5sec to the real session timeout
7793 Allow the session to live 5sec longer before really timing out. This should give
7794 clients some extra time to keep the session active.
7796 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7798 * gst/rtsp-server/rtsp-client.c:
7799 client: replay OK to GET/SET_PARAMETER
7800 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7801 so that we return OK for those requests.
7803 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7805 * gst/rtsp-server/rtsp-media.c:
7806 * gst/rtsp-server/rtsp-media.h:
7807 media: keep track of active transports
7808 Keep track of which transport is active to avoid closing the connection too
7810 Remove the destination transport also when going to NULL.
7811 Print some stats about the SDES and other RTCP messages we receive from the
7814 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7816 * examples/.gitignore:
7817 * examples/Makefile.am:
7818 * examples/test-sdp.c:
7819 example: add SDP relay example
7821 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7823 * gst/rtsp-server/rtsp-media.c:
7824 media: also count active TCP connections
7826 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7828 * gst/rtsp-server/rtsp-media-factory.c:
7829 * gst/rtsp-server/rtsp-media.c:
7830 * gst/rtsp-server/rtsp-media.h:
7831 rtsp: add support for dynamic elements
7832 Add support for dynamic elements.
7833 Don't set live pipelines back to paused.
7835 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7837 * gst/rtsp-server/rtsp-sdp.c:
7838 sdp: don't add encoding name when absent in caps
7840 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7842 * gst/rtsp-server/rtsp-client.c:
7843 client: warn when we can't do RTP-Info
7845 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7847 * gst/rtsp-server/rtsp-media-factory.c:
7848 factory: factor out the stream construction
7850 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7852 * gst/rtsp-server/rtsp-client.c:
7853 client: only add RTP-Info when we have the info
7854 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7857 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7862 === release 0.10.3 ===
7864 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7868 - Fixes a bug where it put the wrong verion in pkgconfig
7869 - Link RTP and RTCP sources
7871 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7873 * gst/rtsp-server/rtsp-media.c:
7874 * gst/rtsp-server/rtsp-media.h:
7875 media: link the RTP udpsrc to the session manager
7876 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7877 shut down when the client sends a packet to open firewalls.
7879 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7881 * pkgconfig/gst-rtsp-server.pc.in:
7882 Don't use hard-coded version number in pkg-config file
7884 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7889 === release 0.10.2 ===
7891 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7896 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7899 * common/m4/.gitignore:
7900 * examples/.gitignore:
7901 * pkgconfig/.gitignore:
7902 add some .gitignore files
7904 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7906 * gst/rtsp-server/rtsp-media.c:
7907 media: seek to key frames
7909 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7911 * gst/rtsp-server/rtsp-media.c:
7912 media: emit the unprepared signal by id
7913 Emit the unprepared signal by id instead of name and set the media as
7916 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7918 * gst/rtsp-server/rtsp-media.c:
7919 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
7921 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7923 * gst/rtsp-server/rtsp-server.c:
7924 Added finalize function to GstRTPSPServer to unref session pool and media mapping
7926 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7928 * bindings/vala/gst-rtsp-server-0.10.vapi:
7929 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7930 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7931 Updated vala bindings
7933 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7935 * gst/rtsp-server/Makefile.am:
7936 * gst/rtsp-server/rtsp-client.c:
7937 * gst/rtsp-server/rtsp-media.c:
7938 server: use appsink and appsrc with the API
7939 Use the appsink/appsrc API instead of the signals for higher
7942 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7944 * examples/test-ogg.c:
7945 tests: set the payload type correctly
7947 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7949 * gst/rtsp-server/rtsp-media-factory.c:
7950 factory: connect to the unprepare signal
7951 Connect to the unprepare signal for non-reusable media so that we can remove
7952 them from the cache.
7954 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7956 * gst/rtsp-server/rtsp-media.c:
7957 * gst/rtsp-server/rtsp-media.h:
7958 media: add signal to notify of unprepare
7960 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7962 * gst/rtsp-server/rtsp-media.c:
7963 * gst/rtsp-server/rtsp-media.h:
7964 media: more work on making the media shared
7965 Add a reusable flag to medias, indicating that they can be reused after a state
7969 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7971 * examples/test-readme.c:
7972 examples: mark the example as shared for testing
7974 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7976 * gst/rtsp-server/rtsp-media.c:
7977 * gst/rtsp-server/rtsp-media.h:
7978 client: support shared media
7979 Always perform the state actions even if the target state of the pipeline is
7980 already correct, we still want to add/remove the transports when we are dealing
7982 Keep a counter of the number of active transports for a media so that we can use
7983 this to perform a state change when needed.
7984 Perform a state change of the pipeline only when the first transport was added
7985 or when there are no active transports.
7987 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7989 * gst/rtsp-server/rtsp-client.c:
7990 client: fix refcounting crasher
7991 Don't need to remove the weak refs in the finalize methods, they are already
7992 removed in the dispose.
7993 Don't register the callback with a DestroyNofity.
7995 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7997 * gst/rtsp-server/rtsp-client.c:
7998 Fix rtsp client refcount management in TCP mode.
7999 Don't unref a client ref we never had. Fixes an unref
8000 of an already-free client object after a client
8001 teardown request for me.
8003 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8005 * gst/rtsp-server/rtsp-session.c:
8006 docs: fix typo in API docs
8008 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8010 * gst/rtsp-server/rtsp-media.c:
8012 Keep the udp sources in playing even if we go to paused. unlock the sources when
8014 Add some more debug info.
8015 Only seek when we need to.
8016 Keep track of the position when we go to paused.
8018 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8020 * gst/rtsp-server/rtsp-client.c:
8021 * gst/rtsp-server/rtsp-media.c:
8022 * gst/rtsp-server/rtsp-media.h:
8023 Add beginnings of seeking.
8024 Parse the Range header and perform a seek on the pipeline for the requested
8025 position. It's disabled currently until I figure out what's going wrong.
8027 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8029 * gst/rtsp-server/rtsp-client.c:
8030 allow pause requests for now.
8033 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8035 * gst/rtsp-server/rtsp-client.c:
8036 Remove weak ref on the session in teardown
8037 We need to remove our weakref from the session when we do a teardown because
8038 else we close the TCP connection prematurely.
8040 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8042 * gst/rtsp-server/rtsp-client.c:
8043 * gst/rtsp-server/rtsp-client.h:
8044 * gst/rtsp-server/rtsp-session-pool.c:
8045 Do some more session cleanup
8046 Make session timeout kill the TCP connection that currently watches the
8048 Remove the client timeout property.
8050 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8052 * gst/rtsp-server/rtsp-client.c:
8053 * gst/rtsp-server/rtsp-client.h:
8054 * gst/rtsp-server/rtsp-media.c:
8055 * gst/rtsp-server/rtsp-media.h:
8056 * gst/rtsp-server/rtsp-server.c:
8057 * gst/rtsp-server/rtsp-session.c:
8058 * gst/rtsp-server/rtsp-session.h:
8060 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8063 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8065 * examples/Makefile.am:
8066 * examples/test-launch.c:
8067 Add example server that takes launch lines
8068 Add an example server that streams any -launch line.
8070 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8072 * examples/test-readme.c:
8073 * gst/rtsp-server/rtsp-client.c:
8074 * gst/rtsp-server/rtsp-media.c:
8075 * gst/rtsp-server/rtsp-media.h:
8076 Add support for live streams
8077 Add support for live streams and ranges
8078 Start on handling TCP data transfer.
8080 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8082 * gst/rtsp-server/rtsp-media.c:
8083 Free the pipeline before other things
8086 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8088 * gst/rtsp-server/rtsp-client.c:
8089 Only free the pending tunnel if there is one
8092 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8094 * gst/rtsp-server/rtsp-client.c:
8095 * gst/rtsp-server/rtsp-client.h:
8096 * gst/rtsp-server/rtsp-media.c:
8097 rtsp-server: Add support for tunneling
8098 Add support for tunneling over HTTP.
8099 Use new connection methods to retrieve the url.
8100 Dispatch messages based on the message type instead of blindly
8101 assuming it's always a request.
8102 Keep track of the watch id so that we can remove it later.
8103 Set the media pipeline to NULL before unreffing the pipeline.
8105 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8107 * gst/rtsp-server/rtsp-client.c:
8108 * gst/rtsp-server/rtsp-client.h:
8109 Fix for channel -> watch rename in gstreamer
8110 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8112 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8114 * gst/rtsp-server/rtsp-client.c:
8115 * gst/rtsp-server/rtsp-client.h:
8117 Use the async RTSP channels instead of spawning a new thread for each client.
8118 If a sessionid is specified in a request, fail if we don't have the session.
8120 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8122 * gst/rtsp-server/rtsp-media.c:
8123 Add better debug info
8124 Add some better debug info.
8126 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8128 * examples/test-video.c:
8130 Add support for session timeouts in the example.
8132 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8134 * gst/rtsp-server/rtsp-session-pool.c:
8135 * gst/rtsp-server/rtsp-session-pool.h:
8136 Pass GTimeVal around for performance reasons
8137 Get the current time only once and pass it around so that sessions don't have to
8138 get the current time anymore.
8139 Add experimental support for a GSource that dispatches when the session needs to
8142 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8144 * gst/rtsp-server/rtsp-session.c:
8145 * gst/rtsp-server/rtsp-session.h:
8146 Add better support for session timeouts
8147 Add a method to request the number of milliseconds when a session will timeout.
8149 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8151 * gst/rtsp-server/rtsp-media.c:
8152 * gst/rtsp-server/rtsp-media.h:
8153 Add suport for RTP manager monitoring
8154 Add the first stage in monitoring the rtp manager.
8155 Make sure we don't update the state to something we don't want.
8157 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8159 * gst/rtsp-server/rtsp-client.c:
8160 Add support for session keepalive
8161 Get and update the session timeout for all requests. get the session as early as
8164 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8166 * gst/rtsp-server/rtsp-media-factory.h:
8167 * gst/rtsp-server/rtsp-media.c:
8168 * gst/rtsp-server/rtsp-media.h:
8169 Handle media bus messages
8170 Handle media bus messages in a custom mainloop and dispatch them to the
8171 RTSPMedia objects. Let the default implementation handle some common messages.
8173 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8175 * gst/rtsp-server/rtsp-client.c:
8176 * gst/rtsp-server/rtsp-session-pool.c:
8177 * gst/rtsp-server/rtsp-session.c:
8178 Some more session timeout handling
8179 Move the session header setting code to a central place so that we always add
8180 the timeout parameter too.
8181 Handle timeouts by running the session cleanup code.
8182 Stop media before cleaning up.
8184 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8186 * gst/rtsp-server/rtsp-client.c:
8187 * gst/rtsp-server/rtsp-client.h:
8188 Add timeout property
8189 Add a timeout property ot the client and make the other properties into GObject
8192 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8194 * gst/rtsp-server/rtsp-session-pool.c:
8195 Use getters and setters in property code
8196 Use the getters and setters for the timeout property instead of locking
8199 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8201 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8203 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8205 * gst/rtsp-server/rtsp-session-pool.c:
8206 * gst/rtsp-server/rtsp-session-pool.h:
8207 * gst/rtsp-server/rtsp-session.c:
8208 * gst/rtsp-server/rtsp-session.h:
8209 Add more timeout stuff
8210 Add method to check if a session is expired.
8211 Add method to perform cleanup on a session pool.
8213 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8215 * gst/rtsp-server/rtsp-client.c:
8216 * gst/rtsp-server/rtsp-session-pool.c:
8217 * gst/rtsp-server/rtsp-session-pool.h:
8218 * gst/rtsp-server/rtsp-session.c:
8219 * gst/rtsp-server/rtsp-session.h:
8220 Add beginnings of session timeouts and limits
8221 Add the timeout value to the Session header for unusual timeout values.
8222 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8223 limit on the amount of retry we do after a sessionid collision.
8224 Add properties to the sessionid and the timeout of a session. Keep track of
8225 creation time and last access time for sessions.
8227 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8229 * gst/rtsp-server/rtsp-client.c:
8230 * gst/rtsp-server/rtsp-media.c:
8231 * gst/rtsp-server/rtsp-media.h:
8232 * gst/rtsp-server/rtsp-sdp.c:
8233 * gst/rtsp-server/rtsp-session-pool.c:
8234 * gst/rtsp-server/rtsp-session.c:
8235 * gst/rtsp-server/rtsp-session.h:
8236 Cleanup of sessions and more
8237 Fix the refcounting of media and sessions in the client. Properly clean up the
8238 session data when the client performs a teardown.
8239 Add Server header to responses.
8240 Allow for multiple uri setups in one session.
8241 Add Range header to the PLAY response and add the range attribute to the SDP
8243 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8244 give the ownership of the sessionid to the session object.
8246 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8248 * gst/rtsp-server/rtsp-server.c:
8249 * gst/rtsp-server/rtsp-server.h:
8251 Rename the 'server_port' variable to simply 'port'.
8253 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8256 * gst/rtsp-server/rtsp-client.c:
8257 * gst/rtsp-server/rtsp-media.c:
8258 * gst/rtsp-server/rtsp-media.h:
8259 * gst/rtsp-server/rtsp-session.c:
8260 * gst/rtsp-server/rtsp-session.h:
8261 Rework the way we handle transports for streams
8262 Make the media accept an array of transports for the streams that we have
8263 configured for the play/pause requests.
8264 Implement server states for a client and its media.
8265 Require 0.10.22.1 (git HEAD) of gstreamer.
8267 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * gst/rtsp-server/rtsp-client.c:
8270 * gst/rtsp-server/rtsp-media-factory.c:
8271 Drop const from functions dealing with urls
8272 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8273 have the right const in them.
8275 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8277 * gst/rtsp-server/rtsp-client.c:
8278 * gst/rtsp-server/rtsp-media.c:
8279 * gst/rtsp-server/rtsp-sdp.c:
8283 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8285 * gst/rtsp-server/rtsp-client.c:
8286 * gst/rtsp-server/rtsp-media-factory.c:
8287 * gst/rtsp-server/rtsp-media.c:
8288 * gst/rtsp-server/rtsp-media.h:
8290 Don't keep a reference to the GstRTSPMedia in the stream.
8291 Free more things when freeing the GstRTSPMedia.
8293 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8296 * gst/rtsp-server/rtsp-media-factory.c:
8297 * gst/rtsp-server/rtsp-media-factory.h:
8298 * gst/rtsp-server/rtsp-media.c:
8299 * gst/rtsp-server/rtsp-media.h:
8300 * gst/rtsp-server/rtsp-server.c:
8301 * gst/rtsp-server/rtsp-server.h:
8302 More docs and small cleanups
8303 Add some more docs and update the README
8304 Cleanup some method names.
8305 Remove an unneeded idx field in the GstRTSPMediaStream
8307 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8310 * examples/Makefile.am:
8311 * examples/test-readme.c:
8312 Add a README and more example code
8313 Add a README file that contains a small introduction on how to use the server
8314 along with the example code explained in the readme.
8316 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8318 * gst/rtsp-server/rtsp-media.c:
8319 * gst/rtsp-server/rtsp-server.c:
8320 Fix some leaks and change default port
8321 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8322 we finished the initial preroll. If we keep them locked, setting the pipeline to
8323 NULL will not stop and clean up the sources correctly.
8324 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8326 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8328 * gst/rtsp-server/rtsp-session.c:
8329 * gst/rtsp-server/rtsp-session.h:
8330 Cleanups to the session object
8331 Remove some unneeded variables in the session state of a stream such as the
8332 owner media and the server transport.
8333 Get the configuration of a media stream in a session based on the media_stream
8334 in the original object instead of our cached index.
8335 Free more data in the finalize method.
8337 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8339 * gst/rtsp-server/rtsp-client.c:
8340 * gst/rtsp-server/rtsp-client.h:
8341 Cleanups and reuse media from DESCRIBE
8342 Handle thread create errors.
8343 Rename some internal methods to better match what they actually do.
8344 Handle misconfiguration of session_pool and media_mapping gracefully.
8345 Cache the DESCRIBE media and uri in the client connection and reuse them when
8346 we receive a SETUP request in the same connection for the same uri.
8347 Cleanup the client connection object.
8349 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8351 * gst/rtsp-server/rtsp-media-factory.c:
8352 * gst/rtsp-server/rtsp-media-factory.h:
8353 * gst/rtsp-server/rtsp-media.c:
8354 * gst/rtsp-server/rtsp-media.h:
8355 Add shared properties to media and factory
8356 Add the shared property to media.
8357 Implement some simple caching in the factory depending on if the media is shared
8360 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8362 * gst/rtsp-server/rtsp-client.c:
8363 Add a little comment
8364 Add some comment about the content-base header.
8366 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8368 * examples/Makefile.am:
8370 * examples/test-mp4.c:
8371 * examples/test-ogg.c:
8372 * examples/test-video.c:
8373 * gst/rtsp-server/Makefile.am:
8374 * gst/rtsp-server/rtsp-client.c:
8375 * gst/rtsp-server/rtsp-client.h:
8376 * gst/rtsp-server/rtsp-media-factory.c:
8377 * gst/rtsp-server/rtsp-media-factory.h:
8378 * gst/rtsp-server/rtsp-media.c:
8379 * gst/rtsp-server/rtsp-media.h:
8380 * gst/rtsp-server/rtsp-sdp.c:
8381 * gst/rtsp-server/rtsp-sdp.h:
8382 * gst/rtsp-server/rtsp-server.c:
8383 * gst/rtsp-server/rtsp-server.h:
8384 * gst/rtsp-server/rtsp-session.c:
8385 * gst/rtsp-server/rtsp-session.h:
8386 Reorganize things, prepare for media sharing
8387 Added various other test server examples
8388 Move the SDP message generation to a separate helper.
8389 Refactor common code for finding the session.
8390 Add content-base for realplayer compatibility
8391 Clean up request uris before processing for better vlc compatibility.
8392 Move prerolling and pipeline construction to the RTSPMedia object.
8393 Use multiudpsink for future pipeline reuse.
8395 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8401 === release 0.10.1 ===
8403 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8409 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8411 * bindings/vala/Makefile.am:
8413 Add more directories and files to the dist.
8415 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8417 * bindings/python/Makefile.am:
8418 * bindings/python/rtspserver.override:
8419 Fixed compile error of python bindings
8421 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8423 * bindings/vala/gst-rtsp-server-0.10.vapi:
8424 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8425 Marked values as nullable accordingly
8427 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8429 * bindings/vala/gst-rtsp-server-0.10.vapi:
8430 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8431 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8432 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8433 Updated Vala bindings
8435 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8437 * gst/rtsp-server/rtsp-client.c:
8438 * gst/rtsp-server/rtsp-media-mapping.c:
8439 * gst/rtsp-server/rtsp-media-mapping.h:
8440 * gst/rtsp-server/rtsp-media.h:
8441 * gst/rtsp-server/rtsp-session-pool.h:
8442 Cleanups and doc updates
8443 Add some more documentation and do some minor cleanups here and there.
8445 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8447 * gst/rtsp-server/rtsp-client.c:
8448 * gst/rtsp-server/rtsp-media-factory.c:
8449 * gst/rtsp-server/rtsp-media-factory.h:
8450 * gst/rtsp-server/rtsp-media.c:
8451 * gst/rtsp-server/rtsp-media.h:
8452 * gst/rtsp-server/rtsp-session.c:
8453 * gst/rtsp-server/rtsp-session.h:
8455 Rename GstRTSPMediaBin to GstRTSPMedia
8456 Parse the request url into a GstRTSPUri object and pass this object to the
8457 various handlers and methods that require the uri.
8459 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8463 Add some more docs and remove some old code from the example.
8465 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-client.c:
8468 Handle state change failures better
8469 Handle state change failures better when changing the state of the pipeline to
8472 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8474 * gst/rtsp-server/rtsp-media-factory.c:
8475 * gst/rtsp-server/rtsp-media-factory.h:
8476 Make element creation more extendible
8477 Add get_element vmethod to the default MediaFactory so that subclasses can just
8478 override that method and still use the default logic for making a MediaBin from
8481 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/Makefile.am:
8485 * gst/rtsp-server/rtsp-client.c:
8486 * gst/rtsp-server/rtsp-client.h:
8487 * gst/rtsp-server/rtsp-media-factory.c:
8488 * gst/rtsp-server/rtsp-media-factory.h:
8489 * gst/rtsp-server/rtsp-media-mapping.c:
8490 * gst/rtsp-server/rtsp-media-mapping.h:
8491 * gst/rtsp-server/rtsp-media.c:
8492 * gst/rtsp-server/rtsp-media.h:
8493 * gst/rtsp-server/rtsp-server.c:
8494 * gst/rtsp-server/rtsp-server.h:
8495 * gst/rtsp-server/rtsp-session.c:
8496 * gst/rtsp-server/rtsp-session.h:
8497 Make the server handle arbitrary pipelines
8498 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8499 The GstMediaBin object has a handle to a bin with elements and to a list of
8500 GstMediaStream objects that this bin produces.
8501 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8502 with methods to register and remove those mappings.
8503 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8504 used by the server instance.
8505 Modify the example application so that it shows how to create custom pipelines
8506 attached to a specific mount point.
8507 Various misc cleanps.
8509 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8511 * gst/rtsp-server/rtsp-server.c:
8512 * gst/rtsp-server/rtsp-server.h:
8513 Allow setting a custom media factory for a server
8515 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8517 * gst/rtsp-server/rtsp-client.c:
8518 * gst/rtsp-server/rtsp-client.h:
8519 Allow setting a custom media factory for a client.
8521 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8523 * gst/rtsp-server/Makefile.am:
8524 Add Makefile entry for the media factory
8526 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8528 * gst/rtsp-server/rtsp-media-factory.c:
8529 * gst/rtsp-server/rtsp-media-factory.h:
8530 Add media factory to map urls to media pipeline objects.
8532 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8534 * gst/rtsp-server/rtsp-media.c:
8535 * gst/rtsp-server/rtsp-media.h:
8536 Add comments. Remove unused field
8538 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-session-pool.c:
8541 * gst/rtsp-server/rtsp-session-pool.h:
8542 Allow custom session pools to override the session id allocation algorithms Add some comments.
8544 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/rtsp-server/rtsp-session.h:
8549 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8551 * gst/rtsp-server/rtsp-client.c:
8552 * gst/rtsp-server/rtsp-client.h:
8553 Move the connection code in one place Add some comments
8555 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8557 * gst/rtsp-server/rtsp-server.c:
8558 * gst/rtsp-server/rtsp-server.h:
8559 Make vmethod to create and accept new clients. Add some docs.
8561 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8563 * gst/rtsp-server/rtsp-server.c:
8564 * gst/rtsp-server/rtsp-server.h:
8565 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8567 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * gst/rtsp-server/rtsp-client.c:
8570 * gst/rtsp-server/rtsp-client.h:
8571 Name the parameters more appropriately.
8573 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8575 * gst/rtsp-server/rtsp-session-pool.c:
8576 Do some more cleanup of the session pool.
8578 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8580 * gst/rtsp-server/Makefile.am:
8581 * gst/rtsp-server/rtsp-client.c:
8582 Check if return value of gst_rtsp_session_get_media is not NULL
8584 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/Makefile.am:
8587 Install rtsp-session and rtsp-session-pool headers
8589 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8594 * bindings/python/Makefile.am:
8595 * bindings/python/arg-types.py:
8596 * bindings/python/codegen/Makefile.am:
8597 * bindings/python/codegen/__init__.py:
8598 * bindings/python/codegen/argtypes.py:
8599 * bindings/python/codegen/code-coverage.py:
8600 * bindings/python/codegen/codegen.py:
8601 * bindings/python/codegen/definitions.py:
8602 * bindings/python/codegen/defsparser.py:
8603 * bindings/python/codegen/docextract.py:
8604 * bindings/python/codegen/docgen.py:
8605 * bindings/python/codegen/fileprefix.override:
8606 * bindings/python/codegen/fileprefixmodule.c:
8607 * bindings/python/codegen/h2def.py:
8608 * bindings/python/codegen/mergedefs.py:
8609 * bindings/python/codegen/mkskel.py:
8610 * bindings/python/codegen/override.py:
8611 * bindings/python/codegen/reversewrapper.py:
8612 * bindings/python/codegen/scmexpr.py:
8613 * bindings/python/rtspserver-types.defs:
8614 * bindings/python/rtspserver.defs:
8615 * bindings/python/rtspserver.override:
8616 * bindings/python/rtspservermodule.c:
8618 Add python bindings.
8620 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8622 * bindings/Makefile.am:
8624 Don't go into python dir when requirements for python bindings are missing
8626 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8628 * bindings/Makefile.am:
8629 * bindings/vala/Makefile.am:
8631 Install Vala bindings if vala is available
8633 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8635 * bindings/vala/gst-rtsp-server-0.10.deps:
8636 * bindings/vala/gst-rtsp-server-0.10.vapi:
8637 * bindings/vala/gst-rtsp-server.vapi:
8638 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8639 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8640 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8641 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8642 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8643 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8644 * bindings/vala/packages/gst-rtsp-server.deps:
8645 * bindings/vala/packages/gst-rtsp-server.excludes:
8646 * bindings/vala/packages/gst-rtsp-server.files:
8647 * bindings/vala/packages/gst-rtsp-server.gi:
8648 * bindings/vala/packages/gst-rtsp-server.metadata:
8649 * bindings/vala/packages/gst-rtsp-server.namespace:
8650 Regenerated Vala bindings
8652 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8654 * bindings/vala/gst-rtsp-server.vapi:
8655 * bindings/vala/packages/gst-rtsp-server.metadata:
8656 Fixed typo in included headers for vala bindings
8658 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * pkgconfig/Makefile.am:
8663 * pkgconfig/gst-rtsp-server.pc.in:
8664 Added pkgconfig file
8666 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8668 * bindings/vala/gst-rtsp-server.vapi:
8669 * bindings/vala/packages/gst-rtsp-server.excludes:
8670 * bindings/vala/packages/gst-rtsp-server.gi:
8671 * bindings/vala/packages/gst-rtsp-server.metadata:
8672 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8674 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8676 * bindings/vala/gst-rtsp-server.vapi:
8677 * bindings/vala/packages/gst-rtsp-server.deps:
8678 * bindings/vala/packages/gst-rtsp-server.files:
8679 * bindings/vala/packages/gst-rtsp-server.gi:
8680 * bindings/vala/packages/gst-rtsp-server.metadata:
8681 * bindings/vala/packages/gst-rtsp-server.namespace:
8684 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8686 * gst/rtsp-server/rtsp-session.c:
8687 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8689 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8691 * examples/Makefile.am:
8692 * gst/rtsp-server/Makefile.am:
8693 Put GStreamer version in library name
8695 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8697 * examples/Makefile.am:
8698 * gst/rtsp-server/Makefile.am:
8699 Fix some issues to pass distcheck
8701 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8703 * gst/rtsp-server/rtsp-server.c:
8704 Added port property to GstRTSPServer class.
8706 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8711 * examples/Makefile.am:
8714 * gst/rtsp-server/Makefile.am:
8715 * gst/rtsp-server/rtsp-client.c:
8716 * gst/rtsp-server/rtsp-client.h:
8717 * gst/rtsp-server/rtsp-media.c:
8718 * gst/rtsp-server/rtsp-media.h:
8719 * gst/rtsp-server/rtsp-server.c:
8720 * gst/rtsp-server/rtsp-server.h:
8721 * gst/rtsp-server/rtsp-session-pool.c:
8722 * gst/rtsp-server/rtsp-session-pool.h:
8723 * gst/rtsp-server/rtsp-session.c:
8724 * gst/rtsp-server/rtsp-session.h:
8727 * src/rtsp-client.c:
8728 * src/rtsp-client.h:
8731 * src/rtsp-server.c:
8732 * src/rtsp-server.h:
8733 * src/rtsp-session-pool.c:
8734 * src/rtsp-session-pool.h:
8735 * src/rtsp-session.c:
8736 * src/rtsp-session.h:
8737 Split in library and example program
8739 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8741 * src/rtsp-client.h:
8742 Removed obsolete variable
8744 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8746 * src/rtsp-client.c:
8747 * src/rtsp-client.h:
8748 Removed pipeline variable GstRTSPClient, because it's only used in one function
8750 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8753 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8755 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8757 * src/rtsp-session.c:
8758 Initialize some more vars.
8760 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8762 * src/rtsp-session.c:
8763 Initialize variable to avoid compiler warning.
8765 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8768 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob