3 2015-12-24 Sebastian Dröge <slomo@coaxion.net>
8 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
11 configure: Make -Bsymbolic check work with clang.
12 Update the -Bsymbolic check with the version glib has. This version
14 https://bugzilla.gnome.org/show_bug.cgi?id=759713
16 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
18 * gst/rtsp-server/rtsp-session-pool.c:
19 rtsp-session-pool: Avoid dollar sign ($) in session ids
20 Live555 in VLC strips off dollar signs and then gets very confused,
21 we don't loose too much entropy by just skipping it.
23 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
25 * gst/rtsp-server/rtsp-address-pool.h:
26 * gst/rtsp-server/rtsp-auth.h:
27 * gst/rtsp-server/rtsp-client.h:
28 * gst/rtsp-server/rtsp-media-factory-uri.h:
29 * gst/rtsp-server/rtsp-media-factory.h:
30 * gst/rtsp-server/rtsp-media.h:
31 * gst/rtsp-server/rtsp-mount-points.h:
32 * gst/rtsp-server/rtsp-permissions.h:
33 * gst/rtsp-server/rtsp-server.h:
34 * gst/rtsp-server/rtsp-session-media.h:
35 * gst/rtsp-server/rtsp-session-pool.h:
36 * gst/rtsp-server/rtsp-session.h:
37 * gst/rtsp-server/rtsp-stream-transport.h:
38 * gst/rtsp-server/rtsp-stream.h:
39 * gst/rtsp-server/rtsp-thread-pool.h:
40 * gst/rtsp-server/rtsp-token.h:
41 rtsp-server: Add g_autoptr() support to all types
42 https://bugzilla.gnome.org/show_bug.cgi?id=754464
44 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
46 * gst/rtsp-server/rtsp-stream.c:
47 rtsp-stream: fixed valgrind error
48 Fixed the valgrind error in unit test. The UDP source created during
49 gst_rtsp_stream_join_bin() was not released while destroying the rtp
51 https://bugzilla.gnome.org/show_bug.cgi?id=759010
53 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
57 Automatic update of common submodule
58 From b319909 to 86e4663
60 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
62 * gst/rtsp-server/rtsp-client.c:
63 rtsp-client: suspend media during setup request
64 SETUP request from clients needs to suspend the media to clear the
65 prerolled buffers. Otherwise it will not affect the prerolled buffer
66 and the prerolled buffers will be incorrect (for example block-size
67 from setup request will not affect the prerolled buffer unless the
69 https://bugzilla.gnome.org/show_bug.cgi?id=758268
71 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
73 * gst/rtsp-server/rtsp-stream.c:
74 rtsp-stream: create stream pipeline based on transport
75 Based on the protocol, create the rtsp stream pipeline. If only TCP or
76 only UDP is set as the transport protocol, it will not add the extra tee
77 or queue element to the pipeline. Both these elements will be added, if
78 it supports both TCP and UDP protocols. This improves the pipeline
79 performance when one protocol is present.
80 https://bugzilla.gnome.org/show_bug.cgi?id=758179
82 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
84 * gst/rtsp-server/rtsp-stream.c:
85 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
86 Adding them when not needed will start some logic inside rtpbin that might be
87 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
88 would start up a rtpjitterbuffer and behave in weird ways.
89 We still set up the UDP sources for RTP receiving for a sender media to be
90 able to receive any packets sent by the client for NAT traversal. They will
91 all go to a fakesink though.
92 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
93 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
94 receive ASYNC_DONE after a seek.
95 https://bugzilla.gnome.org/show_bug.cgi?id=758319
97 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
99 * gst/rtsp-server/rtsp-stream.c:
100 rtsp-stream: Disable multicast loopback for the multicast udp sources too
101 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
102 Previously we were only setting this for sender sockets, which caused looped
103 back packets to be received on Windows if a multicast transport was used.
105 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
107 * examples/test-record-auth.c:
108 * examples/test-record.c:
109 examples: Actually use the provided port in the record examples
111 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
113 * examples/test-record-auth.c:
114 test-record-auth: Add the option to build in TLS support
116 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
118 * examples/test-auth.c:
119 test-auth: Use an 'anonymous' user for unauthenticated default
120 There's a comment on one of the resources that 'user' and 'admin'
121 shouldn't even be able to see it, but they can if the default
122 token is 'admin2', since that gives them access anyway.
124 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
126 * examples/.gitignore:
127 * examples/Makefile.am:
128 * examples/test-record-auth.c:
129 Add test-record-auth example
131 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
133 * gst/rtsp-server/rtsp-client.c:
134 * tests/check/gst/client.c:
135 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
137 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
139 * gst/rtsp-server/rtsp-server.c:
140 rtsp-server: Change the logic so we don't pop a NULL context
141 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
142 will sometimes fail. This call is made before any context is pushed
143 resulting in an attempt to pop a NULL context.
144 https://bugzilla.gnome.org/show_bug.cgi?id=757949
146 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
148 * tests/check/gst/rtspserver.c:
149 rtspserver: Add udp-mcast transport SETUP test
150 Refactor utility functions in the test file so they can handle
151 more than UDP and TCP as lower transport.
152 https://bugzilla.gnome.org/show_bug.cgi?id=756969
154 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
156 * gst/rtsp-server/rtsp-stream.c:
157 rtsp-stream: Always unref return value of gst_object_get_parent()
158 Fixes a leak of a GstBin in the udp-mcast case.
159 https://bugzilla.gnome.org/show_bug.cgi?id=756968
161 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
164 Automatic update of common submodule
165 From b99800a to b319909
167 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
170 Use new GST_ENABLE_EXTRA_CHECKS #define
171 https://bugzilla.gnome.org/show_bug.cgi?id=756870
173 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
176 Automatic update of common submodule
177 From 6babecd to b99800a
179 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
182 Update GLib dependency to 2.40.0
184 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
186 * examples/test-mp4.c:
187 * gst/rtsp-server/rtsp-stream.c:
188 stream: listen to sender ssrc signals
189 https://bugzilla.gnome.org/show_bug.cgi?id=746747
191 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
194 common: update for new suppression
195 Makes check-valgrind pass with glib 2.46
197 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
199 * gst/rtsp-server/rtsp-media.c:
200 rtsp-media: Take reference to media that will be prepared
201 default_prepare() takes a transfer-none reference GstRTSPMedia object.
202 Later on a g_idle_source_new() is created and a pointer to the media
203 object is passed as user data. If the media is freed before the idle
204 source is dispatched the media object pointer is invalid, but the idle
205 source callback expects it to still be valid. To fix this a reference to
206 the media object is taken when registering the source callback function
207 and a corresponding release of the reference is done when the souce is
209 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
211 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
213 * examples/test-launch.c:
214 * examples/test-mp4.c:
215 * examples/test-ogg.c:
216 * examples/test-record.c:
217 * examples/test-uri.c:
218 rtsp-server: Fix memory leaks when context parse fails
219 When g_option_context_parse fails, context and error variables are not getting free'd
220 which results in memory leaks. Free'ing the same.
221 And replacing g_error_free with g_clear_error, which checks if the error being passed
222 is not NULL and sets the variable to NULL on free'ing.
223 https://bugzilla.gnome.org/show_bug.cgi?id=753863
225 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
230 === release 1.6.0 ===
232 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
238 * gst-rtsp-server.doap:
241 === release 1.5.91 ===
243 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
249 * gst-rtsp-server.doap:
252 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
254 * docs/libs/gst-rtsp-server-sections.txt:
255 * gst/rtsp-server/rtsp-stream.c:
256 stream: fix docs for recently-added get/set_buffer_size API
257 https://bugzilla.gnome.org/show_bug.cgi?id=749095
259 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
261 * gst/rtsp-server/rtsp-media.c:
262 rtsp-media: Don't crash on encrypted RTX SDP
263 In parse_keymgmt(), don't mutate the input string that's been passed
264 as const, especially since we might need the original value again if
265 the same key info applies to multiple streams (RTX, for example).
266 https://bugzilla.gnome.org/show_bug.cgi?id=754753
268 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
270 * examples/test-mp4.c:
271 test-mp4: Support filenames with spaces in them. Error out on too few arguments
273 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
275 * examples/test-record.c:
276 test-record: Check parameter count and print out help
277 If no launch pipeline was supplied, print out some help
279 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
281 * gst/rtsp-server/rtsp-media.c:
282 * gst/rtsp-server/rtsp-stream.c:
283 * gst/rtsp-server/rtsp-stream.h:
284 rtsp-stream: Implement UDP buffer size setting.
285 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
287 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
288 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
290 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
292 * gst/rtsp-server/rtsp-media.h:
293 rtsp-media: Fix small typo causing gtk-doc to complain
295 === release 1.5.90 ===
297 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
303 * gst-rtsp-server.doap:
306 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
308 * gst/rtsp-server/rtsp-media-factory.c:
309 media-factory: get port number through gst_rtsp_url_get_port
310 https://bugzilla.gnome.org/show_bug.cgi?id=753473
312 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
314 * tests/check/gst/media.c:
315 media-test: Removing unnecessary assertion
316 https://bugzilla.gnome.org/show_bug.cgi?id=753385
318 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
320 * gst/rtsp-server/rtsp-server.c:
321 Document that source keeps a ref on server until it's destroyed
322 https://bugzilla.gnome.org/show_bug.cgi?id=749227
324 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
326 * tests/check/gst/media.c:
327 media-test: Test for multiple dynamic payload
328 https://bugzilla.gnome.org/show_bug.cgi?id=753385
330 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
332 * gst/rtsp-server/rtsp-media.c:
333 media: Only add fakesink once per pipeline
334 The intention is to prevent going PLAYING state before pads are created.
335 If there was mutilple dynamic payload, it would leak few fakesink and
336 actually prevent from ever reaching playing state.
337 https://bugzilla.gnome.org/show_bug.cgi?id=753385
339 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
341 * gst/rtsp-server/rtsp-media.c:
342 Revert "rtsp-media: Only add 1 fakesink per pipeline"
343 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
345 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
347 * gst/rtsp-server/rtsp-media.c:
348 rtsp-media: Only add 1 fakesink per pipeline
349 There should be only one fakesink per pipeline, not per dynpay. This
350 would lead to element naming clash.
352 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
354 * gst/rtsp-server/rtsp-media.c:
355 rtsp-media: assertion error due to wrong condition check
356 In media to caps function, reserved_keys array is being used for variable i,
357 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
358 changed it to variable j
359 https://bugzilla.gnome.org/show_bug.cgi?id=753009
361 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
363 * gst/rtsp-server/rtsp-media.c:
364 rtsp-media: Strip keys from the fmtp that we use internally in our caps
365 Skip keys from the fmtp, which we already use ourselves for the
366 caps. Some software is adding random things like clock-rate into
367 the fmtp, and we would otherwise here set a string-typed clock-rate
368 in the caps... and thus fail to create valid RTP caps
369 https://bugzilla.gnome.org/show_bug.cgi?id=753009
371 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
373 * gst/rtsp-server/rtsp-thread-pool.c:
374 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
375 https://bugzilla.gnome.org/show_bug.cgi?id=752640
377 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
380 Automatic update of common submodule
381 From f74b2df to 9aed1d7
383 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
388 === release 1.5.2 ===
390 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
396 * gst-rtsp-server.doap:
399 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
401 * gst/rtsp-server/rtsp-client.c:
402 * gst/rtsp-server/rtsp-client.h:
403 * tests/check/gst/client.c:
404 rtsp-client: allow application to decide what requirements are supported
405 Add "check-requirements" signal and vfunc to allow application
406 (and subclasses) to check the requirements.
407 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
408 https://bugzilla.gnome.org/show_bug.cgi?id=749417
410 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
413 Automatic update of common submodule
414 From 6015d26 to f74b2df
416 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
418 * gst/rtsp-server/rtsp-media.c:
419 rtsp-media: Always use real payloader when creating streams
420 A bin that contains the real payloader might be used as payloader. In this
421 case we have to get the real payloader for the various properties it provides.
422 Example use cases for this are bins that payload some media and then have
423 additional elements that add metadata or RTP extension headers to the stream.
424 https://bugzilla.gnome.org/show_bug.cgi?id=750800
426 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
428 * examples/test-netclock-client.c:
429 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
431 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
433 * examples/test-netclock-client.c:
434 * examples/test-netclock.c:
435 test-netclock: Use new ntp-time-source property on rtpbin
436 Select the clock time to be used as NTP time source. This allows proper
437 synchronization between receivers, independent of sharing base times, and just
438 requires them to use the same clock.
440 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
442 * examples/test-netclock-client.c:
443 * examples/test-netclock.c:
444 test-netclock: Setting the same base time on sender and receiver is not necessary
445 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
447 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
449 * gst/rtsp-server/rtsp-stream.c:
450 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
451 https://bugzilla.gnome.org/show_bug.cgi?id=750764
453 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
455 * docs/libs/gst-rtsp-server.types:
456 docs: add missing types
457 https://bugzilla.gnome.org/show_bug.cgi?id=750764
459 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
461 * docs/libs/gst-rtsp-server-sections.txt:
462 docs: add missing apis
463 https://bugzilla.gnome.org/show_bug.cgi?id=750764
465 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
467 * examples/test-netclock-client.c:
468 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
470 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
472 * docs/libs/gst-rtsp-server-sections.txt:
473 * gst/rtsp-server/rtsp-auth.c:
474 * gst/rtsp-server/rtsp-auth.h:
475 GstRTSPAuth: Add client certificate authentication support
476 https://bugzilla.gnome.org/show_bug.cgi?id=750471
478 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
480 * examples/test-netclock-client.c:
481 test-netclock-client: Use new GstClock API to wait for clock synchronization
483 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
485 * examples/test-netclock-client.c:
486 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
487 A mainloop is needed to get glimagesink to display something on OSX, and
488 the source-setup signal just makes things a little bit easier.
490 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
493 Automatic update of common submodule
494 From d9a3353 to 6015d26
496 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
499 Automatic update of common submodule
500 From d37af32 to d9a3353
502 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
505 Automatic update of common submodule
506 From 21ba2e5 to d37af32
508 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
511 Automatic update of common submodule
512 From c408583 to 21ba2e5
514 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
516 * docs/libs/Makefile.am:
517 docs: remove variables that we define in the snippet from common
518 This is syncing our Makefile.am with upstream gtkdoc.
520 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
523 Automatic update of common submodule
524 From 44a3517 to c408583
526 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
531 === release 1.5.1 ===
533 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
539 * gst-rtsp-server.doap:
542 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
544 * gst/rtsp-server/rtsp-client.c:
545 rtsp-client: No flush during Teardown.
546 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
547 backlog is empty it can happen that just a part of a message will be
548 sent and rest is in backlog queue. If then flush during teardown
549 just a part of message will be sent.This can lead to client miss
550 teardown response since it expect to get the last part of message.
551 The flushing during teardown was introduced to fix a deadlock that now
552 is fixed more generally in handle_request by temporary setting backlog
554 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
556 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
558 * tests/check/Makefile.am:
559 tests: Use AM_TESTS_ENVIRONMENT
560 Needed by the new automake test runner and the
561 current version of the common submodule.
563 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
565 * gst/rtsp-server/rtsp-media.h:
566 * gst/rtsp-server/rtsp-stream.h:
567 rtsp-server: Use single-include rtsp header to make sure we get all definitions
569 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
571 * gst/rtsp-server/rtsp-media.c:
572 rtsp-media: Mark some more functions static
574 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
576 * gst/rtsp-server/rtsp-media.c:
577 rtsp-media: Only unblock the media in suspend() when actually changing the state
578 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
580 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
582 * examples/test-video-rtx.c:
583 examples: Use AVPF profile for the RTX example
585 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
587 * gst/rtsp-server/rtsp-sdp.c:
588 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
590 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
592 * gst/rtsp-server/rtsp-stream.c:
593 rtsp-stream: get valid clock-rate from last-sample
594 clock-rate in last-sample's caps is integer, not unsigned.
595 To get this value properly, variable needs to be type-casted to int.
596 https://bugzilla.gnome.org/show_bug.cgi?id=747614
598 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
602 autogen.sh: only run autopoint if gettext requested in configure.ac
603 Not just because there happens to be a po directory.
604 https://bugzilla.gnome.org/show_bug.cgi?id=748058
606 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
609 Revert "configure.ac: uncomment gettext version setup"
610 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
611 We don't need a gettext setup here and there's no po
612 directory either, so no reason why autopoint would be
613 run in the first place.
614 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
616 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
618 * examples/test-multicast.c:
619 * examples/test-multicast2.c:
620 * examples/test-sdp.c:
621 * examples/test-video-rtx.c:
622 * examples/test-video.c:
623 * tests/test-cleanup.c:
624 * tests/test-reuse.c:
625 Fix timeout function signatures across tests and examples
627 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
629 * tests/check/Makefile.am:
630 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
631 Make sure the test environment is set up.
632 https://bugzilla.gnome.org//show_bug.cgi?id=747624
634 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
637 configure: bump automake requirement to 1.14 and autoconf to 2.69
638 This is only required for builds from git, people can still
639 build tarballs if they only have older autotools.
640 https://bugzilla.gnome.org//show_bug.cgi?id=747624
642 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
645 configure.ac: uncomment gettext version setup
646 Fixes autogen.sh. It would run autopoint, which would complain
647 that it could not find the gettext version in configure.ac.
648 https://bugzilla.gnome.org/show_bug.cgi?id=748058
650 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
652 * examples/test-video-rtx.c:
653 test-video-rtx: set exact payload type to PCMA payloader
654 Setting wrong payload type causes failure to do retransmission through audio stream
655 https://bugzilla.gnome.org/show_bug.cgi?id=747839
657 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
659 * gst/rtsp-server/rtsp-media.c:
660 * gst/rtsp-server/rtsp-stream.c:
661 * gst/rtsp-server/rtsp-stream.h:
662 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
663 Because of duplicated g_signal_connect for request-aux-sender signal,
664 wrong stream pointer is passed to the signal handler.
665 Instead of passing each stream, pass stream array and get the relevant stream.
666 https://bugzilla.gnome.org/show_bug.cgi?id=747839
668 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
672 Update autogen.sh to latest version from common
673 Fixes build after aclocal_check etc. helpers have been removed.
675 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
678 Automatic update of common submodule
679 From bc76a8b to c8fb372
681 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
683 * gst/rtsp-server/rtsp-stream.c:
684 rtsp-stream: Limit the queues to 1 buffer
685 We only need them to be able to pre-roll, queueing up more data here
686 is only going to harm latency and memory usage.
688 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
690 * gst/rtsp-server/rtsp-stream.c:
691 rtsp-stream: Update comment and ASCII art to the latest code
692 We have a queue in front of the udpsink too to prevent the pipeline from
695 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
697 * gst/rtsp-server/rtsp-stream.c:
698 rtsp-media: Properly return first rtptime
699 Instead we where returning first GstBuffer timestamp. This would result
700 in clock skew and unwanted behaviour in RTSP playback.
701 https://bugzilla.gnome.org/show_bug.cgi?id=746479
703 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
705 * gst/rtsp-server/rtsp-stream.c:
706 rtsp-stream: Don't leave buffer mapped
707 If the seq is NULL, the RTP buffer was left mapped. We should always
710 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
715 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
717 * gst/rtsp-server/rtsp-media-factory.c:
718 * tests/check/gst/client.c:
719 Fix double semicolons
721 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
723 * gst/rtsp-server/rtsp-stream.c:
724 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
725 This gives more accurate values than asking the payloader. There might be
726 queueing happening between the payloader and the sink.
727 https://bugzilla.gnome.org/show_bug.cgi?id=745704
729 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
731 * gst/rtsp-server/rtsp-media.c:
732 rtsp-media: Don't seek for PLAY if the position will not change
733 https://bugzilla.gnome.org/show_bug.cgi?id=745704
735 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
737 * gst/rtsp-server/rtsp-media.c:
738 rtsp-media: Don't include payload type in the caps for framesize
739 When the sdp media attribute framesize are converted to caps
740 the <payload> should not be included.
741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
742 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
744 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
746 * gst/rtsp-server/rtsp-sdp.c:
747 rtsp-sdp: add payload type to the sdp framesize attribute
748 The sdp framesize attribute is desribed in RFC6064. It is specified
749 for payloading of H263 and has the following form
750 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
751 should be added to the caps in a payloader and the <payload type> should
752 be added by the rtsp-server.
753 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
755 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
757 * examples/test-uri.c:
758 examples: test-uri: fix tainted variable
759 Insignificant but this keeps Coverity happy.
762 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
764 * examples/.gitignore:
765 * examples/Makefile.am:
766 * examples/test-netclock-client.c:
767 * examples/test-netclock.c:
768 examples: Add a simple example of network synch for live streams.
769 An example server and client that works for synchronising live streams
770 only - as it can't support pause/play.
772 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
774 * gst/rtsp-server/rtsp-media-factory.c:
775 * gst/rtsp-server/rtsp-media-factory.h:
776 rtsp-media-factory: Add functions to set/get the media gtype
777 Allow specifying the GType of a GstRtspMedia subclass to create
778 as a simpler way to get the factory to create a custom
779 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
781 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
783 * gst/rtsp-server/rtsp-media.c:
784 rtsp-media: fix double unlock in _get_buffer_size()
785 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
786 because of double g_mutex_unlock () usage.
787 https://bugzilla.gnome.org/show_bug.cgi?id=745434
789 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
791 * gst/rtsp-server/rtsp-session-pool.c:
792 * gst/rtsp-server/rtsp-session.c:
793 * gst/rtsp-server/rtsp-session.h:
794 rtsp-session: Use monotonic time for RTSP session timeout
795 Changed RTSP session timeout handling to monotonic time
796 and deprecating the API for current system time.
797 This fixes timeouts when the system time changes.
798 https://bugzilla.gnome.org/show_bug.cgi?id=743346
800 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
802 * gst/rtsp-server/rtsp-client.c:
803 * gst/rtsp-server/rtsp-media.c:
804 rtsp-client: Only error out in PLAY if seeking actually failed
805 If the media was just not seekable, we continue from whatever position we are
806 and let the client decide if that is what is wanted or not.
807 Only if the actual seek failed, we can't really recover and should error out.
809 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
811 * gst/rtsp-server/rtsp-stream.c:
812 rtsp-stream: Add necessary queues between tee and multiudpsink
813 https://bugzilla.gnome.org/show_bug.cgi?id=744379
815 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
817 * gst/rtsp-server/rtsp-client.c:
818 * gst/rtsp-server/rtsp-media.c:
819 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
820 Instead error out properly the same way as if the SEEKING query already
823 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
825 * gst/rtsp-server/rtsp-stream.h:
826 rtsp-stream: minor code formatting fix
828 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
830 * gst/rtsp-server/rtsp-media.c:
831 rtsp-media: fix logic for collect_streams
832 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
833 all streams it knows if it got any, and can check if the transport mode is OK.
836 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
838 * gst/rtsp-server/rtsp-media.c:
839 rtsp-media: Don't set the transport mode based on what elements we find
840 Just print a warning if the one that was set before disagrees with what
841 elements we found. It must already be set to something before as this
842 function is called after we received the SDP from ANNOUNCE in RECORD mode,
843 and we would reject ANNOUNCE if the RECORD flag was not set.
845 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
847 * tests/check/gst/rtspserver.c:
848 tests: rtspserver: rename shadowed variable
849 We have two different 'sink' variables here,
850 rename one of them for clarity.
852 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
854 * gst/rtsp-server/rtsp-client.c:
855 rtsp-client: fix awkward if clause
857 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
859 * examples/test-uri.c:
860 examples: test-uri: improve uri argument handling and accept file names
861 Print an error if the argument passed is not a URI and can't
862 be converted into one, or no arguments have been provided.
864 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
866 * examples/test-uri.c:
867 examples: test-uri: don't remove mount point after 10 seconds
868 It's very irritating when trying to test stuff repeatedly
869 and serves no real purpose other than showing that it can
872 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
874 * examples/.gitignore:
875 examples: add new test-record to .gitignore
877 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
879 * examples/test-record.c:
880 * gst/rtsp-server/rtsp-client.c:
881 * gst/rtsp-server/rtsp-media-factory.c:
882 * gst/rtsp-server/rtsp-media-factory.h:
883 * gst/rtsp-server/rtsp-media.c:
884 * gst/rtsp-server/rtsp-media.h:
885 * tests/check/gst/rtspserver.c:
886 rtsp-media: Use flags to distinguish between PLAY and RECORD media
888 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
890 * examples/test-record.c:
891 test-record: Set latency for playback-style example to 2s instead of 200ms
893 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
895 * tests/check/gst/rtspserver.c:
896 tests: add some unit tests for ANNOUNCE and RECORD
897 https://bugzilla.gnome.org/show_bug.cgi?id=743175
899 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
901 * gst/rtsp-server/rtsp-client.c:
902 rtsp-client: fix a couple of leaks in handle_announce
904 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
906 * gst/rtsp-server/rtsp-media-factory.c:
907 * gst/rtsp-server/rtsp-media-factory.h:
908 * gst/rtsp-server/rtsp-media.c:
909 * gst/rtsp-server/rtsp-media.h:
910 rtsp-media: Expose latency setting for setting the rtpbin latency
912 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
914 * examples/test-record.c:
915 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
917 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
919 * gst/rtsp-server/rtsp-stream.c:
920 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
922 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
924 * examples/Makefile.am:
925 * examples/test-record.c:
926 * gst/rtsp-server/rtsp-client.c:
927 * gst/rtsp-server/rtsp-client.h:
928 * gst/rtsp-server/rtsp-media-factory.c:
929 * gst/rtsp-server/rtsp-media-factory.h:
930 * gst/rtsp-server/rtsp-media.c:
931 * gst/rtsp-server/rtsp-media.h:
932 * gst/rtsp-server/rtsp-session-media.c:
933 * gst/rtsp-server/rtsp-stream.c:
934 * gst/rtsp-server/rtsp-stream.h:
935 Add initial support for RECORD
936 We currently only support media that is RECORD or PLAY only, not both at once.
937 https://bugzilla.gnome.org/show_bug.cgi?id=743175
939 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
941 * gst/rtsp-server/rtsp-stream.c:
942 rtsp-stream: RTCP and RTP transport cache cookies seperated
943 RTCP packets were not sent because the same tr_cache_cookie was used for
944 both RTP and RTCP. So only one of the tr_cache lists were populated
945 depending on which one was sent first. If the tr_cache list is not
946 populated then no packets can be sent. Most often this happened to be
947 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
948 resulted in both the tr_cache_lists to be populated regardless of which
950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
952 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
954 * gst/rtsp-server/rtsp-stream.c:
955 rtsp-stream: fix false compiler warning
956 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
958 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
960 * gst/rtsp-server/rtsp-client.c:
961 rtsp-client: log interleaved data received
963 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
965 * gst/rtsp-server/rtsp-client.c:
966 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
968 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
970 * gst/rtsp-server/rtsp-client.c:
971 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
973 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
975 * gst/rtsp-server/rtsp-client.c:
976 rtsp-client: Use a random session ID in the SDP
977 RFC4566 Section 5.2 says that it should make the username, session id,
978 nettype, addrtype and unicast address tuple globally unique. Always using
979 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
980 Instead let's create a 64 bit random number, which at least brings us
981 closer to the goal of global uniqueness.
982 https://tools.ietf.org/html/rfc4566#section-5.2
984 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
986 * examples/test-launch.c:
987 * examples/test-mp4.c:
988 * examples/test-ogg.c:
989 * examples/test-uri.c:
990 examples: Don't call gst_init() and gst_get_option_group()
991 The latter calls the former at the appropriate time.
993 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
995 * gst/rtsp-server/rtsp-client.c:
996 rtsp-client: Drop trailing \0 of RTSP DATA messages
997 We add a trailing \0 in GstRTSPConnection to make parsing of
998 string message bodies easier (e.g. the SDP from DESCRIBE) but
999 for actual data this means we have to drop it or otherwise
1000 create invalid data.
1002 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1004 * gst/rtsp-server/rtsp-stream.c:
1005 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1006 Fixes crash when two threads access handle_new_sample() at the same
1007 time, one for RTP, one for RTCP.
1008 Otherwise, when iterating over the transports cache, it might be modified by
1009 another thread at the same time if the transports cookie has changed.
1010 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1012 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1014 * gst/rtsp-server/rtsp-stream.c:
1015 rtsp-stream: Set format=TIME on our app sources for TCP
1017 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1019 * gst/rtsp-server/rtsp-session-pool.c:
1020 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1021 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1022 RFC 2326 states that session IDs may consist of alphanumeric as well as
1023 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1024 Previously the session ID was URI-escaped, this meant that any character
1025 which was not alphanumeric or any of the characters +-._~ would be
1026 percent encoded. While the RFC (surprisingly) mentions that linear white
1027 space in session IDs should be URI-escaped, it does not say anything
1028 about other characters. Moreover no white space is allowed in the
1029 session ID. Finally the percent character which is the result of
1030 URI-escaping is not allowed in a session ID.
1031 So there is no reason to do any URI-escaping, and now it is removed.
1032 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1034 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1037 Automatic update of common submodule
1038 From f2c6b95 to bc76a8b
1040 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1043 Fix 'make check' from top-level directory
1045 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1047 * examples/test-launch.c:
1048 * examples/test-mp4.c:
1049 * examples/test-ogg.c:
1050 * examples/test-uri.c:
1051 examples: Add command-line parsing and take a 'port' argument
1052 This allows users to run multiple servers on different ports for testing.
1053 Only done for examples that actually take arguments and hence are capable of
1054 outputting different streams for each instance on each port.
1055 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1057 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1059 * gst/rtsp-server/rtsp-client.c:
1060 * gst/rtsp-server/rtsp-client.h:
1061 rtsp-client: Add a send_message default signal handler
1062 This allows subclasses to easily hook into the response sending
1063 mechanism without doing everything from a signal, which seems
1064 awkward from subclasses.
1066 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1069 Automatic update of common submodule
1070 From ef1ffdc to f2c6b95
1072 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1076 configure: add --disable-examples switch
1077 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1079 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1081 * examples/.gitignore:
1082 * examples/Makefile.am:
1083 * examples/test-video-rtx.c:
1084 examples: add a retransmisison example implementing RFC4588
1085 Currently only SSRC-multiplexed rtx streams are supported
1087 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1089 * gst/rtsp-server/rtsp-stream.c:
1090 rtsp-stream: Fix some minor memory leaks
1092 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1094 * gst/rtsp-server/rtsp-media.c:
1095 rtsp-media: Some minor cleanup
1097 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1099 * gst/rtsp-server/rtsp-stream.c:
1100 rtsp-stream: Fix compiler warnings
1101 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1102 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1104 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1105 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1108 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1110 * docs/libs/gst-rtsp-server-sections.txt:
1111 * gst/rtsp-server/rtsp-media-factory.c:
1112 * gst/rtsp-server/rtsp-media-factory.h:
1113 * gst/rtsp-server/rtsp-media.c:
1114 * gst/rtsp-server/rtsp-media.h:
1115 * gst/rtsp-server/rtsp-sdp.c:
1116 * gst/rtsp-server/rtsp-stream.c:
1117 * gst/rtsp-server/rtsp-stream.h:
1118 media: implement ssrc-multiplexed retransmission support
1119 based off RFC 4588 and the server-rtpaux example in -good
1121 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1123 * gst/rtsp-server/rtsp-client.c:
1124 * gst/rtsp-server/rtsp-stream-transport.c:
1125 * gst/rtsp-server/rtsp-stream.c:
1126 rtsp: Ref transports in hash table.
1127 Also ref streams for transports.
1128 This solves a crash when reciving a rtcp after teardown but before
1130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1132 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1135 Automatic update of common submodule
1136 From 7bb2bce to ef1ffdc
1138 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1140 * gst/rtsp-server/rtsp-client.c:
1141 client: refactor cleanup of cached media
1143 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1145 * tests/check/gst/client.c:
1147 The session leak is now fixed, lets remove those FIXME comments.
1149 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1151 * tests/check/gst/rtspserver.c:
1152 tests: Test to setup two sessions on one connection
1153 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1155 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1157 * tests/check/gst/rtspserver.c:
1158 tests: Test setup with tcp transport
1159 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1161 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1163 * gst/rtsp-server/rtsp-client.c:
1164 client: Configure transport after creating session media
1165 The default implementation of configure_client_transport() in
1166 rtsp-client uses the session media when it chooses channels for
1167 interleaved traffic.
1168 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1170 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1172 * gst/rtsp-server/rtsp-client.c:
1173 * gst/rtsp-server/rtsp-session-media.c:
1174 client: Stop caching media in client when doing setup
1175 If the media has been managed by a session media, it should not be
1176 cached in the client any longer. The GstRTSPSessionMedia object is now
1177 responsible for unpreparing the GstRTSPMedia object using
1178 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1180 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1182 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1184 * gst/rtsp-server/rtsp-stream.c:
1185 rtsp-stream: unref srtp decoder when leaving bin
1186 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1188 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1190 * gst/rtsp-server/rtsp-client.c:
1191 rtsp-client: mikey memory leaks
1192 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1194 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1197 Automatic update of common submodule
1198 From 84d06cd to 7bb2bce
1200 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1203 Parallelise 'make check-valgrind'
1205 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1208 Automatic update of common submodule
1209 From a8c8939 to 84d06cd
1211 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1214 Automatic update of common submodule
1215 From 36388a1 to a8c8939
1217 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1219 * gst/rtsp-server/rtsp-media.c:
1220 rtsp-media: deactivate media when shutting down from paused
1221 This was only done when going directly from playing.
1222 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1224 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1226 * gst/rtsp-server/rtsp-client.c:
1227 * gst/rtsp-server/rtsp-context.h:
1228 rtsp-client: add stream transport to context
1229 We add the stream transport to the context so we can get the configured
1230 client stream transport in the setup request signal.
1231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1233 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1235 * gst/rtsp-server/rtsp-stream.c:
1236 stream: release lock even not all transports have been removed
1237 We don't want to keep the lock even we return FALSE because not all the
1238 transports have been removed. This could lead into a deadlock.
1239 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1241 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1243 * gst/rtsp-server/rtsp-sdp.c:
1244 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1245 These were renamed in GstRTPBasePayload in 1.0
1247 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1249 * gst/rtsp-server/rtsp-client.c:
1250 client: set session media to NULL without the lock
1251 We need to set session medias to NULL without the client lock otherwise
1252 we can end up in a deadlock if another thread is waiting for the lock
1253 and media unprepare is also waiting for that thread to end.
1254 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1256 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1258 * gst/rtsp-server/rtsp-media.c:
1259 rtsp-media: Set state to UNPREPARING in all cases
1261 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1263 * gst/rtsp-server/rtsp-media.c:
1264 media: set state to unpreparing when unprepare is initiated
1265 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1267 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1269 * gst/rtsp-server/rtsp-client.c:
1270 rtsp-client: Remove backlog limit while processings requests
1271 If the backlog limit is kept two cases of deadlocks may be
1272 encountered when streaming over TCP. Without the backlog
1273 limit this deadlocks can not happen, at the expence of
1275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1277 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1279 * gst/rtsp-server/rtsp-client.c:
1280 rtsp-client: do not free main context before rtsp watch
1281 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1283 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1285 * tests/check/gst/rtspserver.c:
1286 tests: Extend unit test timeout to accomodate for valgrind
1287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1289 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1291 * gst/rtsp-server/rtsp-client.c:
1292 * gst/rtsp-server/rtsp-session.c:
1293 * gst/rtsp-server/rtsp-stream-transport.c:
1294 rtsp-*: Treat sending packets to clients as keepalive
1295 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1296 clients then the client must be reading. This change makes the server
1297 timeout the connection if the client stops reading.
1298 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1300 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1302 * gst/rtsp-server/rtsp-client.c:
1303 rtsp-client: Allow backlog to grow while expiring session
1304 Allow the send backlog in the RTSP watch to grow to unlimited size while
1305 attempting to bring the media pipeline to NULL due to a session
1306 expiring. Without this change the appsink element cannot change state
1307 because it is blocked while rendering data in the new_sample callback.
1308 This callback will block until it has successfully put the data into the
1309 send backlog. There is a chance that the send backlog is full at this
1310 point which means that the callback may block for a long time, possibly
1311 forever. Therefore the media pipeline may also be prevented from
1312 changing state for a long time.
1313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1315 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1317 * gst/rtsp-server/rtsp-client.c:
1318 rtsp-client: Make old compilers happy
1319 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1320 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1322 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1324 * gst/rtsp-server/rtsp-client.c:
1325 client: raise the backlog limits before pausing
1326 We need to raise the backlog limits before pausing the pipeline or else
1327 the appsink might be blocking in the render method in wait_backlog() and
1328 we would deadlock waiting for paused.
1329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1331 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1333 * gst/rtsp-server/rtsp-client.c:
1334 client: make define for the WATCH_BACKLOG
1335 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1337 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1339 * gst/rtsp-server/rtsp-client.c:
1340 client: simplify session transport handling
1341 link/unlink of the transport in a session was done to keep track of all
1342 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1343 that by putting all the TCP transports in a hashtable indexed with the
1345 We also don't need to link/unlink the transports when we pause/resume
1346 the streams. The same effect is already achieved when we pause/play the
1347 media. Indeed, when we pause the media, the transport is removed from
1348 the media and the callbacks will not be called anymore.
1349 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1351 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1353 * gst/rtsp-server/rtsp-stream-transport.c:
1354 * gst/rtsp-server/rtsp-stream-transport.h:
1355 stream-transport: make method to handle received data
1356 Make a method to handle the data received on a channel. It sends the
1357 data to the stream of the transport on the RTP or RTCP pads based on
1360 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1362 * examples/test-mp4.c:
1363 test: add example of dumping RTCP reports
1365 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1367 * gst/rtsp-server/rtsp-media.c:
1368 * gst/rtsp-server/rtsp-stream.c:
1369 * gst/rtsp-server/rtsp-stream.h:
1370 rtsp-media: Make sure that sequence numbers are monotonic after pause
1371 The sequence number is not monotonic for RTP packets after pause. The
1372 reason is basepayloader generates a randon sequence number when the
1373 pipeline goes from ready to pause. With this fix generation of sequence
1374 number will be monotonic when going from pause to play request.
1375 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1377 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1379 * gst/rtsp-server/rtsp-client.c:
1380 rtsp-client: Protect saved clients watch with a mutex
1381 Fixes a crash when close() is called while merging clients
1382 in handle_tunnel(). In that case close() would destroy the
1383 watch while it is still being used in handle_tunnel().
1384 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1386 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1388 * gst/rtsp-server/rtsp-stream.c:
1389 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1391 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1393 * gst/rtsp-server/rtsp-media.c:
1394 * gst/rtsp-server/rtsp-stream.c:
1395 * gst/rtsp-server/rtsp-stream.h:
1396 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1397 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1398 seeking and will always continue counting the time. This leads to
1399 the NPT after a backwards seek to be something completely different
1400 to the actual seek position.
1401 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1403 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1405 * examples/test-appsrc.c:
1406 examples: fix another reference leak
1407 gst_rtsp_media_get_element() returns a new ref.
1409 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1411 * examples/test-appsrc.c:
1412 examples: unref element after usage
1413 gst_bin_get_by_name_recurse_up() returns an element
1414 reference that must be unreffed after usage.
1415 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1417 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1419 * gst/rtsp-server/rtsp-media.c:
1420 signals: Fix copy-pasto in target-state signal offset
1422 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1426 Makefile: Add usage of build-checks step
1427 Allows building checks without running them
1429 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1431 * gst/rtsp-server/rtsp-stream.c:
1432 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1433 When a UDP multicast transport is used it is expected that the server listens
1434 for RTP and RTCP packets on the multicast group with the corresponding port.
1435 Without this we will never get RTCP packets from clients in multicast mode.
1436 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1438 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1443 === release 1.4.0 ===
1445 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1451 * gst-rtsp-server.doap:
1454 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1456 * gst/rtsp-server/rtsp-media.h:
1457 media: correct misspelled words in description
1458 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1460 === release 1.3.91 ===
1462 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1468 * gst-rtsp-server.doap:
1471 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1473 * docs/libs/gst-rtsp-server-sections.txt:
1476 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1478 * gst/rtsp-server/rtsp-server.c:
1479 server: implement client REMOVE filter
1481 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1483 * gst/rtsp-server/rtsp-client.c:
1484 * gst/rtsp-server/rtsp-client.h:
1485 client: expose _close() method
1486 Expose a previously internal close method to close the client
1489 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1491 * gst/rtsp-server/rtsp-session-pool.c:
1492 session-pool: signal session-removed outside of the lock
1493 Release the lock before emiting the session-removed signal.
1495 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1497 * gst/rtsp-server/rtsp-client.c:
1498 * gst/rtsp-server/rtsp-server.c:
1499 * gst/rtsp-server/rtsp-session-pool.c:
1500 * gst/rtsp-server/rtsp-session.c:
1501 * gst/rtsp-server/rtsp-stream.c:
1502 filter: Release lock in filter functions
1503 Release the object lock before calling the filter functions. We need to
1504 keep a cookie to detect when the list changed during the filter
1505 callback. We also keep a hashtable to make sure we only call the filter
1506 function once for each object in case of concurrent modification.
1507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1509 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1511 * gst/rtsp-server/rtsp-client.c:
1512 client: check if watch is set in handle_teardown()
1513 The unit tests run without a watch
1515 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1517 * tests/check/gst/client.c:
1518 client tests: send teardown to cleanup session
1520 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1522 * tests/check/gst/rtspserver.c:
1523 server tests: send teardown to cleanup session
1525 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1527 * gst/rtsp-server/rtsp-client.c:
1528 client: keep ref to client for the session removed handler
1529 This extra ref will be dropped when all client sessions have been
1530 removed. A session is removed when a client sends teardown, closes its
1531 endpoint of the TCP connection or the sessions expires.
1532 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1534 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1536 * gst/rtsp-server/rtsp-client.c:
1537 * gst/rtsp-server/rtsp-session.c:
1538 * tests/check/gst/client.c:
1539 client: manage media in session as a last step
1540 Once we manage a media in a session, we can't unmanage it anymore
1541 without destroying it. Therefore, first check everything before we
1542 manage the media, otherwise if something is wrong we have no way to
1544 If we created a new session and something went wrong, remove the session
1545 again. Fixes a leak in the unit test.
1547 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1549 * examples/test-mp4.c:
1550 * examples/test-ogg.c:
1551 examples: print 'stream ready at url' for mp4 and ogg example
1553 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1555 * gst/rtsp-server/rtsp-client.c:
1556 * gst/rtsp-server/rtsp-sdp.c:
1557 rtsp: fix for MIKEY api change
1559 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1561 * gst/rtsp-server/rtsp-client.c:
1562 client: free watch context only once
1563 The watch context is freed when the source is destroyed. Avoids
1564 a CRITICAL when we try to unref the context twice.
1566 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1568 * gst/rtsp-server/rtsp-client.c:
1571 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1573 * gst/rtsp-server/rtsp-client.c:
1574 client: protect sessions with lock
1575 Protect the list of sessions with the lock.
1576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1578 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1580 * gst/rtsp-server/rtsp-client.c:
1581 Client: keep a ref to the session
1582 Don't just keep a weak ref to the session objects but use a hard ref. We
1583 will be notified when a session is removed from the pool (expired) with
1584 the new session-removed signal.
1585 Don't automatically close the RTSP connection when all the sessions of
1586 a client are removed, a client can continue to operate and it can create
1587 a new session if it wants. If you want to remove the client from the
1588 server, you have to use gst_rtsp_server_client_filter() now.
1589 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1590 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1592 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1594 * gst/rtsp-server/rtsp-session-pool.c:
1595 * gst/rtsp-server/rtsp-session-pool.h:
1596 session-pool: add session-removed signal
1597 Add a signal to be notified when a session is removed from the pool.
1599 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1601 * gst/rtsp-server/Makefile.am:
1602 * gst/rtsp-server/rtsp-server.h:
1603 Make rtsp-server.h a single-include header, use it for G-I
1604 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1606 === release 1.3.90 ===
1608 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1614 * gst-rtsp-server.doap:
1617 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1619 * gst/rtsp-server/rtsp-stream.c:
1620 stream: crypto can be NULL
1622 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1624 * gst/rtsp-server/rtsp-client.c:
1625 * gst/rtsp-server/rtsp-media.c:
1626 * gst/rtsp-server/rtsp-mount-points.c:
1627 introspection: add missing allow-none annotations
1628 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1630 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1632 * gst/rtsp-server/rtsp-address-pool.c:
1633 * gst/rtsp-server/rtsp-media.c:
1634 * gst/rtsp-server/rtsp-session-media.c:
1635 * gst/rtsp-server/rtsp-session-pool.c:
1636 * gst/rtsp-server/rtsp-stream-transport.c:
1637 * gst/rtsp-server/rtsp-stream.c:
1638 * gst/rtsp-server/rtsp-token.c:
1639 introspection: add (nullable) annotations to return values
1640 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1642 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1644 * gst/rtsp-server/rtsp-client.c:
1645 * gst/rtsp-server/rtsp-stream.c:
1646 gi: improve annotations
1647 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1649 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1651 * gst/rtsp-server/rtsp-client.c:
1652 * gst/rtsp-server/rtsp-media-factory.c:
1653 * gst/rtsp-server/rtsp-media.c:
1654 * gst/rtsp-server/rtsp-server.c:
1655 signals: use generic marshal function
1656 Use the generic C marshal function.
1657 Use more explicit type instead of G_TYPE_POINTER
1659 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1661 * gst/rtsp-server/rtsp-context.h:
1662 context: add type macro
1664 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1666 * gst/rtsp-server/rtsp-client.c:
1667 * gst/rtsp-server/rtsp-sdp.c:
1668 * gst/rtsp-server/rtsp-sdp.h:
1669 sdp: hide key length defines
1670 They don't have a namespace.
1672 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1677 === release 1.3.3 ===
1679 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1685 * gst-rtsp-server.doap:
1688 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1690 * gst/rtsp-server/rtsp-client.c:
1691 * gst/rtsp-server/rtsp-sdp.c:
1692 * gst/rtsp-server/rtsp-sdp.h:
1693 mikey: add different key length parameters
1694 Add encryption and authentication key length parameters to MIKEY. For
1695 the encoders, the key lengths are obtained from the cipher and auth
1696 algorithms set in the caps. For the decoders, they are obtained while
1697 parsing the key management from the client.
1698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1700 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1702 * tests/check/gst/stream.c:
1703 stream tests: Make sure we get right multicast address from stream
1704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1706 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1708 * gst/rtsp-server/rtsp-client.c:
1709 client: ref the context until rtsp watch is alive
1710 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1712 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1714 * gst/rtsp-server/rtsp-client.c:
1715 client: Destroy the rtsp watch after connection close
1717 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1719 * gst/rtsp-server/rtsp-media.c:
1720 media: fix confusing comment
1722 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1724 * gst/rtsp-server/rtsp-session.c:
1725 rtsp-session: Timeout in header.
1726 Adding the possbilty to always have timout in header.
1727 This is configurabe with setting "timeout-always-visible".
1728 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1730 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1735 === release 1.3.2 ===
1737 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1744 * gst-rtsp-server.doap:
1747 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1750 Automatic update of common submodule
1751 From 211fa5f to 1f5d3c3
1753 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1755 * gst/rtsp-server/rtsp-client.c:
1756 client: store TCP ports in transport
1757 Store the TCP ports in the transport when we are doing RTSP over TCP.
1758 This way, we can easily get to the ports from the transport.
1759 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1761 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1763 * gst/rtsp-server/rtsp-stream.c:
1764 stream: add signals for new RTP/RTCP encoders
1765 New signals to allow the user to configure the dynamically created
1767 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1769 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1771 * gst/rtsp-server/rtsp-media.c:
1772 * gst/rtsp-server/rtsp-media.h:
1773 media: Make suspend()/unsuspend() virtual
1774 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1776 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1778 * gst/rtsp-server/rtsp-client.c:
1779 client: fix send-message signal marshaller
1780 Use generic marshalling for the send-message signal. It has
1781 two POINTER arguments, not just one.
1782 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1784 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1786 * tests/check/gst/media.c:
1787 tests: add and remove pads only once
1788 In this test we simulate a dynamic pad by watching the caps event.
1789 Because of renegotiation in the base payloader now, this caps is sent
1790 multiple times but we can only deal with 1 invocation, use a variable to
1791 only 'add and remove' the pad once.
1793 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1795 * tests/check/gst/rtspserver.c:
1796 tests: add unit test for correct handling of Require headers
1797 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1799 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1801 * gst/rtsp-server/rtsp-client.c:
1802 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1803 Servers must handle Require headers and must report a failure
1804 if they don't handle any of the Required options, see RFC 2326,
1805 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1806 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1808 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1813 === release 1.3.1 ===
1815 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1821 * gst-rtsp-server.doap:
1824 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1827 Automatic update of common submodule
1828 From bcb1518 to 211fa5f
1830 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1835 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1837 * tests/check/gst/sessionmedia.c:
1838 tests: fix memory leak in sessionmedia unit test
1840 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1842 * gst/rtsp-server/rtsp-client.c:
1843 client: emit a signal before sending a message
1844 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1846 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1848 * gst/rtsp-server/rtsp-client.c:
1849 client: pass context to send_message
1850 Pass the current context to send_message, we will need it later.
1852 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1854 * gst/rtsp-server/rtsp-client.c:
1855 client: fix typo in comment
1857 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1859 * gst/rtsp-server/rtsp-media.c:
1860 media: Do not stop thread twice if default_prepare() fails
1862 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1864 * gst/rtsp-server/rtsp-client.c:
1865 client: set the watch to flushing before going to NULL
1866 First set the watch to flushing so that we unblock any current and
1867 future attempt to send data on the watch, Then set the pipeline to
1869 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1871 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1873 * gst/rtsp-server/rtsp-session-pool.c:
1874 * tests/check/gst/sessionpool.c:
1875 rtsp-session-pool: Fixes annotation
1876 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1877 in the sessionpool test.
1878 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1880 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1882 * gst/rtsp-server/rtsp-media.c:
1883 * gst/rtsp-server/rtsp-media.h:
1884 media: make media_prepare virtual
1885 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1887 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1889 * gst/rtsp-server/rtsp-media.c:
1890 * tests/check/gst/media.c:
1891 media: stop the thread in more error cases
1893 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1895 * gst/rtsp-server/rtsp-media.c:
1896 * tests/check/gst/media.c:
1897 media: allow NULL as the thread
1898 Use the default context whan passing a NULL thread.
1900 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1902 * gst/rtsp-server/rtsp-client.c:
1903 rtsp-client: indent cleanup
1904 Coverity was moaning about unreachable code, and I think it was just
1905 confused by { being before the label. We'll see if it pops up again.
1908 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1910 * gst/rtsp-server/rtsp-client.c:
1911 * gst/rtsp-server/rtsp-media.c:
1912 client: Add drop-backlog property
1913 When we have too many messages queued for a client (currently hardcoded
1914 to 100) we overflow and drop the messages. Add a drop-backlog property
1915 to control this behaviour. Setting this property to FALSE will retry
1916 to send the messages to the client by waiting for more room in the
1918 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1920 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1922 * gst/rtsp-server/rtsp-client.c:
1923 client: support for POST before GET when setting up a tunnel
1925 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1927 * gst/rtsp-server/rtsp-client.c:
1928 client: remove watch of the second client after http tunnel setup
1929 The second client will be freed after the HTTP tunnel has been set up.
1930 Make sure it's RTSP watch is never dispatched again.
1931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1933 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1935 * gst/rtsp-server/rtsp-media.c:
1936 * tests/check/gst/media.c:
1937 media: Make media_prepare() fail if port allocation fails
1938 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1940 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1942 * tests/check/gst/media.c:
1943 media test: cleanup the thread pool in tests
1945 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1947 * gst/rtsp-server/rtsp-media.c:
1948 * tests/check/gst/media.c:
1949 rtsp-media: Unblock blocked streams in unprepare
1950 The streams will be blocked when a live media is prepared.
1951 The streams should be unblocked in gst_rtsp_media_unprepare.
1952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1954 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1956 * gst/rtsp-server/rtsp-media.c:
1957 media: release the state lock when going to NULL
1958 Set our state to UNPREPARING and release the state-lock before
1959 setting the pipeline to the NULL state. This way, any pad-added
1960 callback will be able to take the state-lock and check that we are now
1961 unpreparing instead of deadlocking.
1962 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1964 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1966 * gst/rtsp-server/rtsp-media.c:
1967 media: protect status with lock
1968 Make sure we only update the status with the lock.
1970 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1972 * gst/rtsp-server/rtsp-client.c:
1973 * gst/rtsp-server/rtsp-sdp.c:
1974 rtsp: update for MIKEY API changes
1976 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1978 * gst/rtsp-server/rtsp-client.c:
1979 client: parse the mikey response from the client
1980 Parse the mikey response from the client and update the policy for
1983 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1985 * gst/rtsp-server/rtsp-stream.c:
1986 * gst/rtsp-server/rtsp-stream.h:
1987 stream: add method to set crypto info
1988 Make a method to configure the crypto information of a stream.
1989 Set udpsrc in READY instead of PAUSED so that we can configure caps
1992 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1994 * gst/rtsp-server/rtsp-client.c:
1995 client: cleanup error paths
1997 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1999 * gst/rtsp-server/rtsp-media.c:
2002 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2004 * examples/test-video.c:
2005 test: enable SRTP only on RTSPS
2006 We only want to enable SRTP when doing rtsp over TLS so that we can
2007 exchange the keys in a secure way.
2009 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2011 * examples/test-video.c:
2012 test: print an error on failure
2014 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2017 * examples/test-video.c:
2018 * gst/rtsp-server/rtsp-sdp.c:
2019 * gst/rtsp-server/rtsp-stream.c:
2020 * tests/check/Makefile.am:
2021 stream: add SRTP support
2022 Install srtp encoder and decoder elements in rtpbin
2025 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2027 * tests/check/Makefile.am:
2028 * tests/check/gst/sessionpool.c:
2029 tests: Add unit tests for sessionpool
2030 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2032 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2034 * tests/check/gst/threadpool.c:
2035 tests: Improve code coverage of rtsp-threadpool tests
2036 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2038 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2040 * tests/check/gst/sessionmedia.c:
2041 tests: Improve code coverage for rtsp-session-media
2042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2044 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2046 gobject-introspection: Add annotations to support language bindings
2047 In addition a few cosmetic changes:
2048 * Adjust the order of arguments
2049 * Fix typo: occured -> occurred
2050 * Fix indentation after Return:-clauses
2051 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2053 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2055 * gst/rtsp-server/rtsp-stream.c:
2056 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2057 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2059 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2061 * gst/rtsp-server/rtsp-stream.c:
2062 stream: take caps after the session manager
2063 Take the caps for the SDP after they leave the rtpbin so that we can
2064 also get the properties added by rtpbin elements.
2066 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2068 * gst/rtsp-server/rtsp-stream.c:
2069 stream: release lock while pushing out packets
2070 Keep a cache of the transports and use this to iterate the transport
2071 while pushing packets. This allows us to release the lock early.
2072 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2074 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2076 * gst/rtsp-server/rtsp-client.c:
2077 * gst/rtsp-server/rtsp-client.h:
2078 rtsp-client: vmethod for modifying tunnel GET response
2079 Add a vmethod tunnel_http_response where the response to the HTTP GET
2080 for tunneled connections can be modified.
2081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2083 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2085 * gst/rtsp-server/rtsp-sdp.c:
2086 sdp: make 1 media line per profile
2087 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2088 line in the SDP for each profile. The client is then supposed to pick
2089 one of the profiles in the SETUP request. Because the m= lines have the
2090 same pt, the client also knows that only 1 option is possible.
2092 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2094 * gst/rtsp-server/rtsp-media-factory.c:
2095 * gst/rtsp-server/rtsp-media-factory.h:
2096 * gst/rtsp-server/rtsp-media.c:
2097 factory: add profile property and pass to media and streams
2099 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2101 * examples/test-multicast.c:
2102 * gst/rtsp-server/rtsp-sdp.c:
2103 sdp: pass multicast connection for multicast-only stream
2104 Pass the multicast address of the stream in the connection info in the
2105 SDP so that clients try a multicast connection first.
2106 Only allow multicast connections in the test-multicast example. Also
2107 increase the TTL a little.
2109 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2112 .gitignore: Ignore gcov intermediate files
2113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2115 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2117 * gst/rtsp-server/rtsp-stream.c:
2118 stream: release some locks in error cases
2120 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2122 docs: Enable and fix gtk-doc warnings
2123 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2124 * addresspool/mediafactory: Add missing annotation colon
2125 * stream: Annotate return value
2126 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2128 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2131 Automatic update of common submodule
2132 From fe1672e to bcb1518
2134 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2137 Automatic update of common submodule
2138 From 1a07da9 to fe1672e
2140 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2142 * examples/Makefile.am:
2143 examples: use LDADD for libs instead of LDFLAGS
2145 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2148 configure: make sure releases are in .doap file
2150 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2152 * examples/test-cgroups.c:
2153 examples: test-cgroups: don't put code with side effects into g_assert()
2154 The g_assert() might get compiled out with the right
2155 compiler/preprocessor flags.
2157 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2159 * examples/.gitignore:
2160 examples: add cgroup test binary to .gitignore
2162 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2164 * examples/test-cgroups.c:
2165 examples: fix cgroup test build
2166 Fixes build failure caused by compiler warning:
2167 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2169 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2172 .gitignore: ignore temp files created in the course of 'make check'
2174 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2176 * gst/rtsp-server/rtsp-media.c:
2177 rtsp-media: don't loose frames handling new PLAY request
2178 If client supplied a range check if the range specifies the start point.
2179 If not, then do an accurate seek to the current position. If a start
2180 point was specified do do a key unit seek to make sure the streaming
2181 starts with decodeable frames.
2182 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2184 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2186 * gst/rtsp-server/rtsp-media.c:
2187 Revert "media: only flush when setting a new start position"
2188 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2189 We need to do the flush in all cases, demuxer block currently for
2192 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2194 * gst/rtsp-server/rtsp-media.c:
2195 media: only flush when setting a new start position
2196 Only flush the pipeline when we change the start position with
2198 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2200 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2202 * gst/rtsp-server/rtsp-stream.c:
2203 stream: set ttl-mc before adding the socket
2204 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2205 never be set on socket.
2206 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2208 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2210 * gst/rtsp-server/rtsp-media.c:
2211 media: stop thread if media is already prepared
2212 in gst_rtsp_media_prepare() the thread is not used if media is already
2213 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2215 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2217 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2220 build: Ship gst-rtsp-server.doap file
2222 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2224 * tests/check/gst/rtspserver.c:
2225 tests: Fix another compiler warning with gcc
2227 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2229 * gst/rtsp-server/rtsp-client.c:
2230 * gst/rtsp-server/rtsp-mount-points.c:
2231 * gst/rtsp-server/rtsp-stream.c:
2232 * tests/check/gst/client.c:
2233 rtsp-server: Fix lots of compiler warnings with clang
2235 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2238 * gst-rtsp-server.doap:
2239 * tests/Makefile.am:
2240 configure: Synchronise with the configure scripts of the other modules
2242 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2245 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2247 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2249 * gst/rtsp-server/rtsp-media.c:
2250 * gst/rtsp-server/rtsp-stream.c:
2251 Revert "rtsp-server: support build against last stable release"
2252 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2253 Let us require 1.2.3 now, which is going to be released in a few
2256 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2258 * gst/rtsp-server/rtsp-session-media.c:
2259 * gst/rtsp-server/rtsp-stream-transport.c:
2260 session: improve RTP-Info
2261 Ignore streams that can't generate RTP-Info instead of failing.
2262 Don't return the empty string when all streams are unconfigured but
2263 return NULL so that we don't generate and empty RTP-Info header.
2264 Improve docs a little.
2266 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2268 * gst/rtsp-server/rtsp-session-media.c:
2269 Don't free rtpinfo GString when it is NULL
2270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2272 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2274 * gst/rtsp-server/rtsp-media.c:
2275 media: only set keyframe flag when modifying start
2276 Only set the keyframe flag when we modify the start position. The
2277 keyframe flag should probably be ignored when no change is requested but
2278 until we can claim this is all documented properly and all demuxer
2279 implement this, avoid setting the flag.
2280 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2282 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2284 * gst/rtsp-server/rtsp-thread-pool.c:
2285 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2288 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2290 * gst/rtsp-server/rtsp-stream.c:
2291 stream: handle NULL seqnum and rtptime arguments
2293 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2295 * gst/rtsp-server/rtsp-thread-pool.c:
2296 * tests/check/gst/threadpool.c:
2297 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2298 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2300 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2302 * gst/rtsp-server/rtsp-stream.c:
2303 stream: add fallback for missing stats property
2304 Use a fallback when the payloader does not have a stats property
2305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2307 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2310 Automatic update of common submodule
2311 From f7bc1c3 to 1a07da9
2313 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2315 * gst/rtsp-server/rtsp-stream.c:
2316 stream: don't leak stats structure
2317 Don't leak the stats structure and deal with NULL stats.
2319 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2321 * gst/rtsp-server/rtsp-stream.c:
2322 stream: Get rtpinfo properties atomically from payloader
2323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2325 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2327 * gst/rtsp-server/rtsp-media.c:
2328 media: refactor state change functions and signals
2329 Make functions to set the target state and the pipeline state and emit
2330 the signals from those functions.
2332 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2334 * gst/rtsp-server/rtsp-media.c:
2335 * gst/rtsp-server/rtsp-media.h:
2336 media: add signal to notify of pending state changes
2338 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2340 * gst/rtsp-server/rtsp-media.c:
2341 * gst/rtsp-server/rtsp-stream.c:
2342 rtsp-server: support build against last stable release
2343 Until 1.2.3 is out with the new get_type function and we
2346 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2348 * gst/rtsp-server/rtsp-stream.c:
2349 stream: fix compilation
2351 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2353 * gst/rtsp-server/rtsp-media.c:
2354 * gst/rtsp-server/rtsp-media.h:
2355 * gst/rtsp-server/rtsp-stream.c:
2356 * gst/rtsp-server/rtsp-stream.h:
2357 stream: add property to configure profiles
2359 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2361 * gst/rtsp-server/rtsp-client.c:
2362 client: let stream check supported transport
2363 Delegate the check if a transport is allowed to the stream.
2364 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2366 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2368 * gst/rtsp-server/rtsp-stream.c:
2369 * gst/rtsp-server/rtsp-stream.h:
2370 stream: add method to check supported transport
2371 Add a method to check if a transport is supported
2373 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2376 configure.ac: Only check for gstreamer-check, not check
2377 We include check in gstreamer-check since quite some time now.
2379 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2381 * gst/rtsp-server/rtsp-session-media.c:
2382 * gst/rtsp-server/rtsp-stream-transport.c:
2383 * gst/rtsp-server/rtsp-stream.c:
2384 * gst/rtsp-server/rtsp-stream.h:
2385 stream: return clock-rate from get_rtpinfo
2386 And use it to correct the rtptime to the requested start-time.
2387 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2389 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2391 * gst/rtsp-server/rtsp-session-media.c:
2392 * gst/rtsp-server/rtsp-stream-transport.c:
2393 * gst/rtsp-server/rtsp-stream-transport.h:
2394 session-media: calculate start-time
2396 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2398 * gst/rtsp-server/rtsp-stream-transport.c:
2399 * gst/rtsp-server/rtsp-stream.c:
2400 * gst/rtsp-server/rtsp-stream.h:
2401 stream: also return the running-time
2402 Return the running-time in the rtpinfo as well.
2404 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2406 * gst/rtsp-server/rtsp-client.c:
2407 * gst/rtsp-server/rtsp-session-media.c:
2408 * gst/rtsp-server/rtsp-session-media.h:
2409 * gst/rtsp-server/rtsp-stream-transport.c:
2410 * gst/rtsp-server/rtsp-stream-transport.h:
2411 session-media: let the session-media make the RTPInfo
2412 Add method to create the RTPInfo for a stream-transport.
2413 Add method to create the RTPInfo for all stream-transports in a
2415 Use the session-media RTPInfo code in client. This allows us to refactor
2416 another method to link the TCP callbacks.
2418 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2420 mount-points: sort sequence before g_sequence_lookup
2421 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2422 sort sequence if dirty, otherwise lookup will fail.
2423 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2425 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2428 configure: rename package from gst-rtsp to gst-rtsp-server
2429 To match git module name and avoid confusion with the
2430 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2432 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2435 configure: bump core/base/good requirement to 1.2.0
2436 Bump to released stable version and make implicit
2437 requirements explicit.
2439 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2444 Fix broken gettext setup which is not used anyway
2446 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2449 Automatic update of common submodule
2450 From dbedaa0 to d48bed3
2452 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2454 * gst/rtsp-server/rtsp-client.c:
2455 * gst/rtsp-server/rtsp-media.c:
2456 * gst/rtsp-server/rtsp-media.h:
2457 media: add setup_sdp vmethod
2458 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2459 gst_rtsp_media_setup_sdp.
2460 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2462 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2464 * gst/rtsp-server/rtsp-stream.c:
2465 rtsp-stream: Check return value of sscanf
2466 streamid is only valid if sscanf matched something.
2468 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2470 * gst/rtsp-server/rtsp-client.c:
2471 rtsp-client: Fix iteration
2472 Wouldn't even enter the code block otherwise (i++ was used as the check
2473 and not the postfix).
2475 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2477 * gst/rtsp-server/rtsp-client.c:
2478 * gst/rtsp-server/rtsp-client.h:
2479 client: add vmethod to configure media and streams
2480 Implement a vmethod that can be used to configure the media and the
2481 streams based on the current context. Handle the blocksize handling in
2482 the default handler.
2483 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2485 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2488 Make git ignore more unit test binaries
2490 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2492 * gst/rtsp-server/rtsp-address-pool.h:
2493 * gst/rtsp-server/rtsp-auth.h:
2494 * gst/rtsp-server/rtsp-client.h:
2495 * gst/rtsp-server/rtsp-context.h:
2496 * gst/rtsp-server/rtsp-media-factory-uri.h:
2497 * gst/rtsp-server/rtsp-media-factory.h:
2498 * gst/rtsp-server/rtsp-media.h:
2499 * gst/rtsp-server/rtsp-mount-points.h:
2500 * gst/rtsp-server/rtsp-server.h:
2501 * gst/rtsp-server/rtsp-session-media.h:
2502 * gst/rtsp-server/rtsp-session-pool.h:
2503 * gst/rtsp-server/rtsp-session.h:
2504 * gst/rtsp-server/rtsp-stream-transport.h:
2505 * gst/rtsp-server/rtsp-stream.h:
2506 * gst/rtsp-server/rtsp-thread-pool.h:
2507 * gst/rtsp-server/rtsp-token.h:
2508 rtsp-server: add padding to many public structures
2509 Not mini objects though, since they are not subclassable
2510 anyway, nor kept on the stack or inlined in a structure.
2512 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2514 media: add new create_rtpbin vmethod
2515 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2516 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2518 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2520 * tests/check/gst/media.c:
2521 tests: fix memory leak, free test's thread pool
2522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2524 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2526 * gst/rtsp-server/rtsp-stream-transport.c:
2527 stream-transport: free url in finalize
2529 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2531 * gst/rtsp-server/rtsp-media.c:
2532 media: also do state change in suspended state
2534 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2536 * gst/rtsp-server/rtsp-client.c:
2537 * gst/rtsp-server/rtsp-media.c:
2538 media: also handle prepare and range in suspended state
2539 When we are suspended, we are already prepared.
2540 We can get the range in the suspended state.
2542 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2544 * tests/check/Makefile.am:
2545 * tests/check/gst/sessionmedia.c:
2546 check: add test for uri in setup
2547 Added unit tests for the new functionality in GstRTSPStreamTransport.
2548 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2550 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2552 * gst/rtsp-server/rtsp-client.c:
2553 client: store setup uri and use in PLAY response
2554 Store the uri used when doing the setup and use that in the PLAY
2556 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2558 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2560 * gst/rtsp-server/rtsp-stream-transport.c:
2561 * gst/rtsp-server/rtsp-stream-transport.h:
2562 stream-transport: add method to get/set url
2564 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2566 * gst/rtsp-server/rtsp-client.c:
2567 client: suspend after SDP and unsuspend before PLAYING
2568 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2571 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2573 * gst/rtsp-server/rtsp-media-factory.c:
2574 * gst/rtsp-server/rtsp-media-factory.h:
2575 * gst/rtsp-server/rtsp-media.c:
2576 * gst/rtsp-server/rtsp-media.h:
2577 * gst/rtsp-server/rtsp-session-media.c:
2578 * gst/rtsp-server/rtsp-session.c:
2579 * tests/check/gst/media.c:
2580 * tests/check/gst/mediafactory.c:
2581 media: add suspend modes
2582 Add support for different suspend modes. The stream is suspended right after
2583 producing the SDP and after PAUSE. Different suspend modes are available that
2584 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2585 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2586 state and RESET will bring the pipeline to the NULL state.
2587 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2588 this means that the pipeline needs to be prerolled again.
2589 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2590 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2592 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2594 * gst/rtsp-server/rtsp-media.c:
2595 media: start live streams in blocked state
2596 Start live streams in the blocked state and make them preroll using the
2597 messages. This ensure that no data is played by the sink until we explicitly
2598 unblock the stream right before going to PLAYING.
2599 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2601 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2603 * gst/rtsp-server/rtsp-media.c:
2604 media: refactor starting and waiting for preroll
2605 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2606 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2608 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2610 * gst/rtsp-server/rtsp-stream.c:
2611 * gst/rtsp-server/rtsp-stream.h:
2612 stream: add API to block streams
2613 Add an API to block on the streams and make it post a message.
2614 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2615 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2617 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2619 * docs/libs/Makefile.am:
2620 docs: Specify the override file
2621 Even if it's empty (for now) it avoids make distcheck complaining
2623 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2625 * gst/rtsp-server/rtsp-media.c:
2626 media: move default implementations to where they are used
2628 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2630 * gst/rtsp-server/rtsp-media.c:
2631 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2632 We need to take the state_lock when calling this method.
2634 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2636 * gst/rtsp-server/rtsp-media.c:
2637 media: handle add-added on non-bins too
2638 Handle dynamic payloaders that are not bins, as used in the unit-test.
2640 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2642 * gst/rtsp-server/rtsp-media-factory.c:
2643 * gst/rtsp-server/rtsp-media-factory.h:
2644 * gst/rtsp-server/rtsp-media.c:
2645 rtsp-media/-factory: Fix request pad name comments
2646 These must be escaped for gtk-doc to parse the comments without warnings.
2648 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2650 rtsp-media: remove transports if media is in error status
2651 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2652 trying to change to GST_STATE_NULL and media is in error status, we
2653 remove all transports.
2654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2656 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2658 * gst/rtsp-server/rtsp-media.c:
2659 rtsp-media: use element metadata to find payloader
2660 Use the element metadata to find the payloader instead of checking
2662 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2664 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2666 rtsp-stream: add getter for payload type
2667 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2668 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2669 element and create the stream with this one instead of the dynpay%d
2671 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2673 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2675 * gst/rtsp-server/rtsp-client.c:
2676 * gst/rtsp-server/rtsp-context.h:
2677 * gst/rtsp-server/rtsp-media.c:
2678 * gst/rtsp-server/rtsp-mount-points.c:
2679 * gst/rtsp-server/rtsp-server.c:
2680 * gst/rtsp-server/rtsp-token.c:
2681 rtsp-*: Refer to NULL as a constant in comments
2683 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2685 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2687 rtsp-*: Fix type name typos in comments
2688 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2689 * rtsp-auth: Refer to part of constant name as text
2690 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2691 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2692 * rtsp-stream: Fix typo when refering to GstBin
2693 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2695 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2698 * docs/libs/gst-rtsp-server-docs.sgml:
2699 * docs/libs/gst-rtsp-server-sections.txt:
2700 docs: Improve documentation
2701 * Include annotation-glossary to quiet gtk-doc
2702 * Rename remaining ClientState -> Context
2703 * Rename object hierarchy file
2704 * Remove stale chapter references
2705 * Add missing function and object references
2706 * Include missing GstRTSPAddressPoolResult
2707 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2709 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2711 * gst/rtsp-server/rtsp-client.c:
2712 * gst/rtsp-server/rtsp-server.c:
2713 * gst/rtsp-server/rtsp-session-pool.c:
2714 * gst/rtsp-server/rtsp-session.c:
2715 * gst/rtsp-server/rtsp-stream.c:
2716 rtsp-server: sprinkle some allow-none annotations for g-i
2718 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2720 * gst/rtsp-server/rtsp-stream.c:
2721 * gst/rtsp-server/rtsp-stream.h:
2722 stream: add method to filter transports
2723 Add a method to safely iterate and collect the stream transports
2724 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2726 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2728 * gst/rtsp-server/rtsp-client.c:
2729 * gst/rtsp-server/rtsp-server.c:
2730 * gst/rtsp-server/rtsp-session-pool.c:
2731 * gst/rtsp-server/rtsp-session.c:
2732 rtsp: allow NULL func in filters
2733 Passing a null function make the filters return a list of
2736 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2738 * gst/rtsp-server/rtsp-address-pool.c:
2739 * tests/check/gst/addresspool.c:
2740 address-pool: fix address increment
2741 Use a guint instead of guint8 to increment the address. It's still not
2742 completely correct because a guint might not be able to hold the complete
2743 address range, but that's an enhacement for later.
2744 Add unit test to test improved behaviour.
2745 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2747 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2749 * gst/rtsp-server/rtsp-client.c:
2750 * tests/check/gst/client.c:
2751 client: allow absolute path in requests
2752 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2754 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2756 * gst/rtsp-server/rtsp-client.c:
2757 * gst/rtsp-server/rtsp-client.h:
2758 client: make make_path_from_uri a vmethod
2760 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2762 * docs/libs/gst-rtsp-server-sections.txt:
2763 * gst/rtsp-server/rtsp-stream.c:
2764 * gst/rtsp-server/rtsp-stream.h:
2765 * tests/check/Makefile.am:
2766 * tests/check/gst/stream.c:
2767 stream: Add functions to get rtp and rtcp sockets
2768 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2770 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2772 * gst/rtsp-server/rtsp-context.c:
2773 * gst/rtsp-server/rtsp-context.h:
2774 context: defing a GType for the context
2775 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2777 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2779 * gst/rtsp-server/Makefile.am:
2780 * gst/rtsp-server/rtsp-auth.c:
2781 * gst/rtsp-server/rtsp-context.c:
2782 * gst/rtsp-server/rtsp-media.c:
2783 * gst/rtsp-server/rtsp-mount-points.c:
2784 * gst/rtsp-server/rtsp-server.h:
2785 * gst/rtsp-server/rtsp-session-media.c:
2786 * gst/rtsp-server/rtsp-session.c:
2787 * gst/rtsp-server/rtsp-stream.c:
2788 Fixed several GIR warnings
2790 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2792 * gst/rtsp-server/rtsp-auth.c:
2795 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2797 * tests/check/Makefile.am:
2798 * tests/check/gst/token.c:
2799 tests: Add unit tests for token
2800 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2802 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2804 * gst/rtsp-server/rtsp-token.c:
2805 token: Validate args for gst_rtsp_token_is_allowed
2806 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2808 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2810 * gst/rtsp-server/rtsp-token.c:
2811 token: Fix bug when creating empty token
2812 We always want to have a valid GstStructure in the token.
2813 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2815 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2817 * gst/rtsp-server/rtsp-thread-pool.c:
2818 thread-pool: avoid race in shutdown
2819 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2820 don't actually stop the mainloop ever. Solve this race by adding an idle source
2821 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2822 if quit was called before we started it.
2824 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2826 * tests/check/Makefile.am:
2827 * tests/check/gst/permissions.c:
2828 tests: Add unit tests for permissions
2829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2831 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2833 * tests/check/gst/mediafactory.c:
2834 tests: Test mediafactory permissions
2835 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2837 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2839 * gst/rtsp-server/rtsp-permissions.c:
2840 permissions: Fix refcounting when adding/removing roles
2841 Previously a role that was removed was unreffed twice, and when
2842 replacing an existing role the replaced role was freed while still being
2843 referenced. Both bugs are now fixed.
2844 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2846 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2848 * tests/check/gst/media.c:
2849 * tests/check/gst/mediafactory.c:
2850 * tests/check/gst/rtspserver.c:
2851 tests: Check gst_rtsp_url_parse return value
2852 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2854 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2857 Automatic update of common submodule
2858 From 865aa20 to dbedaa0
2860 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2862 * gst/rtsp-server/rtsp-server.c:
2863 rtsp-server: Fix socket leak
2864 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2866 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2868 * gst/rtsp-server/rtsp-session-pool.c:
2869 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2870 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2872 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2874 * examples/.gitignore:
2875 * examples/test-video.c:
2876 examples: fix compilation when WITH_AUTH is defined
2877 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2879 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2882 gitignore: Add new test binary
2884 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2886 * tests/check/Makefile.am:
2887 * tests/check/gst/threadpool.c:
2888 thread-pool: Add unit test for the thread pools
2889 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2891 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2893 * gst/rtsp-server/rtsp-thread-pool.c:
2894 thread-pool: Fix thread leak when reusing threads
2895 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2897 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2899 * gst/rtsp-server/rtsp-server.c:
2900 * tests/check/gst/rtspserver.c:
2901 tests: fixed racy behavior in rtspserver tests
2902 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2904 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2906 * tests/check/gst/addresspool.c:
2907 tests: Improve address pool unit tests
2908 Add a range with mixed IPV4 and IPV6 addresses to pool.
2909 Get an IPV4 address from an IPV6-only pool.
2910 Get an IPV6 address from an IPV4-only pool.
2911 Reserve a IPV6 address from an IPV4-only pool.
2912 Check for unicast addresses in multicast-only pool.
2913 Check for unicast addresses in uni-/multicast-mixed pool.
2914 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2916 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2918 * gst/rtsp-server/rtsp-client.c:
2919 client: append query string in PAUSE/PLAY/TEARDOWN as well
2921 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2923 * gst/rtsp-server/rtsp-client.c:
2924 client: Add query to control path
2925 If the SETUP url contains a query it must be appended to the control
2926 path so that it matches any already created stream in the media. The
2927 query will also be appended to the session media path.
2929 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2931 * gst/rtsp-server/rtsp-media.c:
2932 rtsp-media: remove old line
2934 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2936 * gst/rtsp-server/rtsp-stream.c:
2937 stream: Correct control comparison
2938 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2940 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2942 * gst/rtsp-server/rtsp-media.c:
2943 media: Check dynamically if the pipeline supports seeking
2944 We should not depend on whether or not the pipeline state change
2945 returned NO_PREROLL or not. A media could dynamically change its
2946 element and switch from seekable to non seekable so it's best to test
2947 the seekable nature of the pipeline dynamically when we try to do a seek.
2949 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2951 * gst/rtsp-server/rtsp-media.c:
2952 media: Return FALSE if seeking is not supported
2954 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2956 * gst/rtsp-server/rtsp-media.c:
2957 rtsp-media: don't seek accurate by default
2958 Accurate seeking is perhaps a little overkill in the most common situation and
2959 causes some formats (mp3) over slow media to seek extremely slowly.
2961 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2963 * tests/check/gst/rtspserver.c:
2964 tests: fix unit test
2965 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2967 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2969 * gst/rtsp-server/rtsp-client.c:
2970 client: Reply 400 if media cannot be constructed
2971 Reply 400 Bad Request instead of 503 Service Unavailable if media
2972 cannot be constructed in SETUP.
2973 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2975 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2977 * gst/rtsp-server/rtsp-client.c:
2978 client: Send setup reply once only
2979 If find_media() failed in handle_setup_request() two replies was sent.
2980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2982 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2985 Automatic update of common submodule
2986 From 6b03ba7 to 865aa20
2988 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2990 * gst/rtsp-server/rtsp-server.c:
2991 server: Emit client-connected signal earlier
2992 Emit client-connected before the client ref is given to a GSource,
2993 otherwise client-connected can be emitted after the client object has
2996 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2998 * gst/rtsp-server/rtsp-address-pool.c:
2999 * gst/rtsp-server/rtsp-address-pool.h:
3000 * gst/rtsp-server/rtsp-stream.c:
3001 * tests/check/gst/addresspool.c:
3002 addresspool: return reason of failure
3003 Let gst_rtsp_address_pool_reserve_address() return the reason why
3004 the address could not be reserved.
3005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3007 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3010 autogen.sh: Sync behaviour with other GStreamer modules
3011 Allows building from outside of tree amongst other things
3013 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3016 Automatic update of common submodule
3017 From b613661 to 6b03ba7
3019 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3022 Automatic update of common submodule
3023 From 74a6857 to b613661
3025 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3028 Automatic update of common submodule
3029 From 01a7a46 to 74a6857
3031 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3033 * gst/rtsp-server/rtsp-client.c:
3034 client: Do not read beyond end of path string
3035 If the setup was done without a control url, make sure we don't try to read the
3036 non-existing control string and crash.
3038 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3040 * gst/rtsp-server/rtsp-client.c:
3041 client: Fix RTPInfo header
3042 Refactor the method to make the content_base.
3043 Use the content-base and the control url to construct the RTPInfo
3046 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3048 * gst/rtsp-server/rtsp-client.c:
3049 client: map url to path only in describe
3050 Only map the request url to a path in the DESCRIBE method. The SDP then
3051 contains the base and control urls that should be used to SETUP/PAUSE/
3052 PLAY/TEARDOWN the media.
3054 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3056 * gst/rtsp-server/rtsp-client.c:
3057 Revert "client: map URL to path in requests"
3058 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3059 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3060 contains the base and control urls which are used in the SETUP, PLAY,
3061 PAUSE and TEARDOWN requests.
3063 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3065 * gst/rtsp-server/rtsp-client.c:
3066 client: map URL to path in requests
3068 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3070 * gst/rtsp-server/rtsp-client.c:
3071 * gst/rtsp-server/rtsp-mount-points.c:
3072 * gst/rtsp-server/rtsp-mount-points.h:
3073 mount-points: make vmethod to make path from uri
3074 Make a vmethod to transform an url into a path. The path is then used to lookup
3075 the factory. This makes it possible to also use other bits of the url, such as
3076 the query parameters, to locate the factory.
3078 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3080 * gst/rtsp-server/rtsp-thread-pool.c:
3081 * gst/rtsp-server/rtsp-thread-pool.h:
3082 thread-pool: Add cleanup to wait for the threadpool to finish
3083 Also fix race condition if two threads are asking for the first
3084 thread from the thread pool at once. This would case two internal
3085 GThreadPools to be created.
3086 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3088 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3090 * gst/rtsp-server/rtsp-client.c:
3091 * tests/check/gst/client.c:
3092 client: free threadpool
3093 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3095 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3097 * tests/check/gst/mountpoints.c:
3098 mountpoints tests: unref matched factories
3099 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3101 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3103 * tests/check/gst/media.c:
3104 media tests: unref thread pool and caps
3105 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3107 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3109 * gst/rtsp-server/rtsp-auth.c:
3110 * gst/rtsp-server/rtsp-media-factory.c:
3111 * gst/rtsp-server/rtsp-media.c:
3112 auth, media, media-factory: unref permissions
3113 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3115 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3117 * examples/Makefile.am:
3118 Makefile: add rule for appsrc example
3120 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3122 * examples/test-appsrc.c:
3123 tests: add appsrc example
3124 Add an example on how to use appsrc to feed the server pipeline with data.
3126 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3128 * gst/rtsp-server/rtsp-client.c:
3129 rtsp-client: remove query part from content-base string
3130 Make sure that after the control url has been resolved, it's
3131 not a part of the query-string.
3132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3134 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3136 * gst/rtsp-server/rtsp-client.c:
3137 client: don't check url in response
3138 There is no url or method in the response to check
3140 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3142 * gst/rtsp-server/rtsp-client.c:
3143 * gst/rtsp-server/rtsp-client.h:
3144 Add handle-response signal for when we receive a GET_PARAMETER response
3146 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3148 * gst/rtsp-server/rtsp-server.c:
3149 Fix gst_rtsp_server_client_filter, using wrong variable type
3151 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3153 * gst/rtsp-server/rtsp-media-factory-uri.c:
3154 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3155 For AAC we need to check for framed=true instead of parsed=true.
3156 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3158 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3160 * gst/rtsp-server/rtsp-stream.c:
3161 stream: optimize pipeline for protocols
3162 When TCP is not an allowed protocol for the stream, avoid creating the
3163 appsrc/appsink/queue and tee elements.
3165 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3167 * gst/rtsp-server/rtsp-media.c:
3168 media: set protocols on streams
3170 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3172 * gst/rtsp-server/rtsp-client.c:
3173 client: use protocols supported by stream
3175 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3177 * gst/rtsp-server/rtsp-media-factory.c:
3178 * gst/rtsp-server/rtsp-media.c:
3179 * gst/rtsp-server/rtsp-stream.c:
3180 media-factory: allow all protocols
3182 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3184 * gst/rtsp-server/rtsp-media.c:
3185 media: configure protocols in new streams
3187 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3189 * gst/rtsp-server/rtsp-stream.c:
3190 * gst/rtsp-server/rtsp-stream.h:
3191 stream: add protocols property
3193 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3195 * gst/rtsp-server/rtsp-media.c:
3196 rtsp-media: send state in "new-state" signal
3197 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3199 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3202 build: add subdir-objects to AM_INIT_AUTOMAKE
3203 Fixes warnings with automake 1.14
3204 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3206 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3208 * docs/libs/gst-rtsp-server-sections.txt:
3209 * gst/rtsp-server/rtsp-client.c:
3210 * gst/rtsp-server/rtsp-server.c:
3211 * gst/rtsp-server/rtsp-server.h:
3212 server: add method to iterate clients of server
3214 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3216 * gst/rtsp-server/rtsp-media.c:
3217 * gst/rtsp-server/rtsp-media.h:
3218 Add vmethod for rtsp-media subclass to access rtpbin
3220 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3222 * gst/rtsp-server/rtsp-client.h:
3223 small documentation fix
3225 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3227 * gst/rtsp-server/rtsp-client.c:
3228 Do not take range header if range is invalid
3230 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3232 * docs/libs/gst-rtsp-server-sections.txt:
3233 * gst/rtsp-server/rtsp-media.c:
3234 media: add docs for new method
3236 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3238 * gst/rtsp-server/rtsp-media.c:
3239 * gst/rtsp-server/rtsp-media.h:
3240 Add API to rtsp-media set the pipeline's state
3242 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3244 * gst/rtsp-server/rtsp-media.c:
3245 Update current position/duration when gst_rtsp_media_get_range_string is called
3247 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3249 * examples/test-cgroups.c:
3250 tests: add some more docs
3252 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3254 * examples/test-cgroups.c:
3255 * gst/rtsp-server/Makefile.am:
3256 * gst/rtsp-server/rtsp-auth.c:
3257 * gst/rtsp-server/rtsp-auth.h:
3258 * gst/rtsp-server/rtsp-client.c:
3259 * gst/rtsp-server/rtsp-client.h:
3260 * gst/rtsp-server/rtsp-context.c:
3261 * gst/rtsp-server/rtsp-context.h:
3262 * gst/rtsp-server/rtsp-params.c:
3263 * gst/rtsp-server/rtsp-params.h:
3264 * gst/rtsp-server/rtsp-server.c:
3265 * gst/rtsp-server/rtsp-thread-pool.c:
3266 * gst/rtsp-server/rtsp-thread-pool.h:
3267 * tests/check/gst/client.c:
3268 ClientState -> Context
3269 Rename the clientstate to context and put the code in a separate file.
3271 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3273 * examples/test-auth.c:
3274 * gst/rtsp-server/rtsp-auth.c:
3275 * gst/rtsp-server/rtsp-auth.h:
3276 auth: add support for default token
3277 The default token is used when the user is not authenticated and can be used to
3278 give minimal permissions.
3280 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3282 * examples/test-auth.c:
3283 * gst/rtsp-server/rtsp-auth.c:
3284 auth: use defines when possible
3286 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3288 * gst/rtsp-server/rtsp-address-pool.c:
3289 address-pool: improve docs
3291 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3293 * gst/rtsp-server/rtsp-permissions.c:
3294 permissions: add the role to the copy
3296 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3298 * gst/rtsp-server/rtsp-permissions.c:
3299 permissions: Also copy the roles
3301 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3303 * gst/rtsp-server/rtsp-permissions.c:
3304 permissions: Make it build
3306 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3308 * gst/rtsp-server/rtsp-address-pool.h:
3311 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3313 * docs/libs/gst-rtsp-server-sections.txt:
3314 * gst/rtsp-server/rtsp-auth.c:
3315 * gst/rtsp-server/rtsp-auth.h:
3316 * gst/rtsp-server/rtsp-media.c:
3317 * gst/rtsp-server/rtsp-session-media.c:
3318 * gst/rtsp-server/rtsp-stream-transport.c:
3319 * gst/rtsp-server/rtsp-stream-transport.h:
3320 * gst/rtsp-server/rtsp-stream.c:
3321 * tests/check/gst/client.c:
3324 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3326 * docs/libs/gst-rtsp-server-sections.txt:
3327 * gst/rtsp-server/rtsp-address-pool.c:
3328 * gst/rtsp-server/rtsp-address-pool.h:
3329 * tests/check/gst/addresspool.c:
3330 * tests/check/gst/rtspserver.c:
3331 address-pool: cleanups
3332 Remove redundant method, improve docs.
3334 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3336 * docs/libs/gst-rtsp-server-sections.txt:
3337 * gst/rtsp-server/rtsp-auth.h:
3338 * gst/rtsp-server/rtsp-permissions.c:
3339 * gst/rtsp-server/rtsp-permissions.h:
3340 * gst/rtsp-server/rtsp-token.c:
3343 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3345 * gst/rtsp-server/rtsp-permissions.c:
3346 permissions: implement _remove_role
3348 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3350 * gst/rtsp-server/rtsp-permissions.c:
3351 permissions: update docs
3353 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3355 * tests/check/gst/client.c:
3356 tests: simplify tests
3357 Client settings are now disabled by default so we don't need an auth
3358 module to disable them.
3360 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3362 * gst/rtsp-server/rtsp-auth.c:
3363 auth: add default authorizations
3364 When no auth module is specified, use our table of defaults to look up the
3365 default value of the check instead of always allowing everything. This was
3366 we can disallow client settings by default.
3368 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3371 README: update readme
3373 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3375 * gst/rtsp-server/rtsp-thread-pool.c:
3376 * gst/rtsp-server/rtsp-thread-pool.h:
3377 thread-pool: add more docs
3379 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3381 * gst/rtsp-server/rtsp-thread-pool.c:
3382 * gst/rtsp-server/rtsp-thread-pool.h:
3383 thread-pool: fix race in thread reuse
3384 If we try to reuse a thread right after we made it stop, we end up using a
3385 stopped thread. Catch this case and only reuse threads that are not stopping.
3387 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3389 * gst/rtsp-server/rtsp-server.c:
3390 server: add small debug
3392 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3394 * tests/check/gst/client.c:
3396 Add some permissions to media so we can use the auth and enable
3399 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3401 * gst/rtsp-server/rtsp-client.c:
3402 client: support pushed context in handle_request
3403 If we already have a pushed state, reuse it and add our own things. This makes
3404 it easier to write tests.
3406 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3408 * gst/rtsp-server/rtsp-auth.c:
3409 auth: don't auth on methods
3410 Don't authorize on methods anymore but on the resources that we
3411 try to access, this is more flexible.
3412 Move the authorization checks to where they are needed and let the
3413 check return the response on error.
3415 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3417 * gst/rtsp-server/rtsp-mount-points.c:
3418 mount-points: add some debug
3420 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3422 * tests/check/gst/client.c:
3423 tests: almost fix test
3425 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3427 * gst/rtsp-server/rtsp-auth.c:
3428 * gst/rtsp-server/rtsp-auth.h:
3429 * gst/rtsp-server/rtsp-client.c:
3430 * gst/rtsp-server/rtsp-client.h:
3431 * gst/rtsp-server/rtsp-server.c:
3432 * gst/rtsp-server/rtsp-server.h:
3433 auth: let the auth module check client_settings
3434 Let the auth module decide if client settings are allowed for the
3437 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3439 * gst/rtsp-server/rtsp-token.c:
3440 * gst/rtsp-server/rtsp-token.h:
3441 token: add method to check boolean permission
3443 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3445 * examples/test-auth.c:
3446 * examples/test-cgroups.c:
3447 * gst/rtsp-server/rtsp-token.c:
3448 * gst/rtsp-server/rtsp-token.h:
3449 token: simplify token constructor
3450 Use variable arguments to make easier API.
3452 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3454 * examples/test-auth.c:
3455 * examples/test-cgroups.c:
3456 * gst/rtsp-server/rtsp-media-factory.c:
3457 * gst/rtsp-server/rtsp-media-factory.h:
3458 media-factory: add convenience API for factory
3460 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3462 * examples/test-auth.c:
3463 * examples/test-cgroups.c:
3464 * gst/rtsp-server/rtsp-permissions.c:
3465 * gst/rtsp-server/rtsp-permissions.h:
3466 permissions: simplify API a little
3467 Avoid passing GstStructure in the add_role method, use varargs instead
3468 to construct the structure behind the scenes. We can then also use the
3469 structure name as the role and simplify some more logic.
3471 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3473 * gst/rtsp-server/rtsp-auth.c:
3476 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3478 * gst/rtsp-server/rtsp-auth.c:
3479 * gst/rtsp-server/rtsp-auth.h:
3480 * gst/rtsp-server/rtsp-client.c:
3481 auth: handle unauthorized response
3482 Move handling of the unauthorized response to the auth module, it can add
3483 the appropriate headers to request authorization for the required method
3484 much better than the client.
3486 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3488 * gst/rtsp-server/rtsp-client.c:
3489 * gst/rtsp-server/rtsp-client.h:
3490 client: allow for sending any message, not only requests
3491 Change the _send_request() method to _send_message() so that we
3492 can both send requests and replies.
3494 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3496 * docs/libs/gst-rtsp-server-sections.txt:
3497 * gst/rtsp-server/rtsp-server.h:
3500 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3502 * examples/test-video.c:
3503 * gst/rtsp-server/rtsp-auth.c:
3504 * gst/rtsp-server/rtsp-auth.h:
3505 * gst/rtsp-server/rtsp-server.c:
3506 * gst/rtsp-server/rtsp-server.h:
3507 auth: move TLS handling to auth module
3508 Remove the TLS settings on the server and move it to the auth module because
3509 that is where security related bits go.
3511 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3513 * gst/rtsp-server/rtsp-client.c:
3514 * gst/rtsp-server/rtsp-client.h:
3515 client: add state push/pop
3517 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3519 * gst/rtsp-server/rtsp-client.c:
3520 * gst/rtsp-server/rtsp-client.h:
3521 client: add connection to state
3523 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3525 * gst/rtsp-server/rtsp-mount-points.c:
3526 mount-points: fix debug
3528 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3530 * tests/check/gst/media.c:
3531 tests: fix media test
3533 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3535 * gst/rtsp-server/rtsp-thread-pool.c:
3536 thread-pool: we don't require a state
3538 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3540 * gst/rtsp-server/rtsp-server.c:
3541 server: let context ref the server
3542 So that we don't risk losing the server object early anc crash.
3544 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3546 * tests/check/gst/client.c:
3547 tests: fix client test
3549 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3552 * docs/libs/gst-rtsp-server-docs.sgml:
3553 * docs/libs/gst-rtsp-server-sections.txt:
3554 * gst/rtsp-server/rtsp-address-pool.c:
3555 * gst/rtsp-server/rtsp-auth.c:
3556 * gst/rtsp-server/rtsp-client.c:
3557 * gst/rtsp-server/rtsp-client.h:
3558 * gst/rtsp-server/rtsp-media-factory-uri.c:
3559 * gst/rtsp-server/rtsp-media-factory.c:
3560 * gst/rtsp-server/rtsp-media-factory.h:
3561 * gst/rtsp-server/rtsp-media.c:
3562 * gst/rtsp-server/rtsp-mount-points.c:
3563 * gst/rtsp-server/rtsp-params.c:
3564 * gst/rtsp-server/rtsp-permissions.c:
3565 * gst/rtsp-server/rtsp-sdp.c:
3566 * gst/rtsp-server/rtsp-server.c:
3567 * gst/rtsp-server/rtsp-server.h:
3568 * gst/rtsp-server/rtsp-session-media.c:
3569 * gst/rtsp-server/rtsp-session-pool.c:
3570 * gst/rtsp-server/rtsp-session.c:
3571 * gst/rtsp-server/rtsp-stream-transport.c:
3572 * gst/rtsp-server/rtsp-stream.c:
3573 * gst/rtsp-server/rtsp-thread-pool.c:
3574 * gst/rtsp-server/rtsp-token.c:
3577 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3579 * gst/rtsp-server/rtsp-session-pool.c:
3580 * gst/rtsp-server/rtsp-session-pool.h:
3581 session-pool: make vmethod to create a session
3582 Make a vmethod to create a sessions so that subclasses can create
3583 custom session objects
3585 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3587 * gst/rtsp-server/rtsp-auth.c:
3588 * gst/rtsp-server/rtsp-media-factory.h:
3589 * gst/rtsp-server/rtsp-media.h:
3590 * gst/rtsp-server/rtsp-mount-points.h:
3591 * gst/rtsp-server/rtsp-session-pool.h:
3592 * gst/rtsp-server/rtsp-stream.h:
3595 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3597 * docs/libs/gst-rtsp-server-docs.sgml:
3598 * docs/libs/gst-rtsp-server-sections.txt:
3599 * gst/rtsp-server/rtsp-address-pool.c:
3600 * gst/rtsp-server/rtsp-address-pool.h:
3601 * gst/rtsp-server/rtsp-auth.c:
3602 * gst/rtsp-server/rtsp-client.h:
3603 * gst/rtsp-server/rtsp-media-factory.h:
3604 * gst/rtsp-server/rtsp-media.c:
3605 * gst/rtsp-server/rtsp-media.h:
3606 * gst/rtsp-server/rtsp-permissions.c:
3607 * gst/rtsp-server/rtsp-permissions.h:
3608 * gst/rtsp-server/rtsp-server.h:
3609 * gst/rtsp-server/rtsp-session-media.c:
3610 * gst/rtsp-server/rtsp-session-media.h:
3611 * gst/rtsp-server/rtsp-session-pool.h:
3612 * gst/rtsp-server/rtsp-session.h:
3613 * gst/rtsp-server/rtsp-stream-transport.h:
3614 * gst/rtsp-server/rtsp-stream.c:
3615 * gst/rtsp-server/rtsp-thread-pool.h:
3618 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3621 * examples/Makefile.am:
3622 configure: compile cgroup example conditionally
3623 Only compile the cgroup example when we have libcgroup
3625 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3628 * examples/Makefile.am:
3629 * examples/test-cgroups.c:
3630 examples: add cgroups example
3632 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3634 * tests/check/gst/rtspserver.c:
3635 tests: fix compilation
3637 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3639 * gst/rtsp-server/rtsp-thread-pool.c:
3640 thread-pool: fix vmethod invocation
3642 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3644 * gst/rtsp-server/rtsp-thread-pool.c:
3645 * gst/rtsp-server/rtsp-thread-pool.h:
3646 thread-pool: store thread type in thread
3648 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3650 * gst/rtsp-server/rtsp-client.c:
3651 client: pass thread from pool to media _prepare
3652 Get a thread from the configured threadpool and pass it to the prepare method of
3655 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3657 * gst/rtsp-server/rtsp-media.c:
3658 * gst/rtsp-server/rtsp-media.h:
3659 media: Accept a thread in _prepare
3660 Remove out own threadpool handling and use the provided thread and
3661 maincontext for the bus messages and the state changes.
3663 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3665 * gst/rtsp-server/rtsp-server.c:
3666 server: configure client thread pool
3668 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3670 * gst/rtsp-server/rtsp-client.c:
3671 * gst/rtsp-server/rtsp-client.h:
3672 client: add method to configure thread pool
3674 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3676 * gst/rtsp-server/rtsp-client.h:
3677 * gst/rtsp-server/rtsp-server.c:
3678 * gst/rtsp-server/rtsp-server.h:
3679 server: use thread pool
3680 Use the thread pool instead of doing our own thing.
3682 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3684 * gst/rtsp-server/Makefile.am:
3685 * gst/rtsp-server/rtsp-thread-pool.c:
3686 * gst/rtsp-server/rtsp-thread-pool.h:
3687 thread-pool: add object to manage threads
3688 Add an object to manage the client and media threads.
3690 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3692 * gst/rtsp-server/rtsp-auth.c:
3693 auth: debug authorization check
3695 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3697 * gst/rtsp-server/rtsp-media.c:
3698 media: start media pipeline in context
3699 Start the media pipeline in the provided context (or our default one
3700 when NULL). This makes sure that we run the bus thread in this context and that
3701 all media threads are children of this context.
3703 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3705 * gst/rtsp-server/rtsp-media-factory.c:
3706 factory: pass permissions to media by default
3708 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3710 * examples/test-auth.c:
3711 test: add permissions to auth test
3712 Ass some permissions to the media factory in the test.
3714 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * gst/rtsp-server/rtsp-auth.c:
3717 * gst/rtsp-server/rtsp-auth.h:
3718 * gst/rtsp-server/rtsp-client.c:
3719 auth: simplify auth checks
3720 Remove client from methods, it's now in the state
3721 Perform the check specified by the string, use the information from the
3722 thread local context.
3724 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3726 * gst/rtsp-server/rtsp-client.c:
3727 * gst/rtsp-server/rtsp-client.h:
3728 client: add state to current thread
3729 Add the client to the ClientState object.
3730 Place the ClientState on the current thread.
3732 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3734 * gst/rtsp-server/rtsp-media-factory.c:
3735 * gst/rtsp-server/rtsp-media-factory.h:
3736 * gst/rtsp-server/rtsp-media.c:
3737 * gst/rtsp-server/rtsp-media.h:
3738 media: make it possible to set permissions
3739 Make it possible to set permissions on media and media factory objects
3741 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3743 * gst/rtsp-server/Makefile.am:
3744 * gst/rtsp-server/rtsp-permissions.c:
3745 * gst/rtsp-server/rtsp-permissions.h:
3746 permissions: add permissions object
3747 Add a mini object to store permissions based on a role.
3749 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3751 * examples/test-auth.c:
3752 * gst/rtsp-server/rtsp-auth.c:
3753 * gst/rtsp-server/rtsp-auth.h:
3754 * gst/rtsp-server/rtsp-client.c:
3755 auth: add auth checks
3756 Add an enum with auth checks and implement the checks in the auth object.
3757 Perform the checks from the client.
3759 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3761 * examples/test-auth.c:
3762 * gst/rtsp-server/rtsp-auth.c:
3763 * gst/rtsp-server/rtsp-auth.h:
3764 * gst/rtsp-server/rtsp-client.h:
3765 auth: use the token after authentication
3766 After we authenticated a user, keep the Token around in the state.
3768 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3770 * gst/rtsp-server/rtsp-client.c:
3771 * gst/rtsp-server/rtsp-media.c:
3772 * gst/rtsp-server/rtsp-media.h:
3773 * tests/check/gst/media.c:
3774 media: add optional context for bus messages
3775 Add an optional mainloop to _prepare that will handle the bus messages instead
3776 of always using the shared mainloop.
3778 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * gst/rtsp-server/Makefile.am:
3781 * gst/rtsp-server/rtsp-token.c:
3782 * gst/rtsp-server/rtsp-token.h:
3783 token: add authorization token
3784 Add a simply miniobject that contains the authorizations. The object contains a
3785 GstStructure that hold all authorization fields. When a user is authenticated,
3786 the auth module will create a Token for the user. The token is then used to
3787 check what operations the user is allowed to do and various other configuration
3790 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3792 * examples/test-auth.c:
3793 * gst/rtsp-server/rtsp-auth.c:
3794 * gst/rtsp-server/rtsp-auth.h:
3795 * gst/rtsp-server/rtsp-client.c:
3796 * gst/rtsp-server/rtsp-client.h:
3797 * gst/rtsp-server/rtsp-media-factory.c:
3798 * gst/rtsp-server/rtsp-media-factory.h:
3799 * gst/rtsp-server/rtsp-media.c:
3800 * gst/rtsp-server/rtsp-media.h:
3801 auth: remove auth from media and factory
3802 Remove the auth object from media and factory. We want to have the RTSPClient
3803 authenticate and authorize resources, there is no need to place another auth
3804 manager on the media/factory.
3806 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3808 * examples/test-auth.c:
3809 * gst/rtsp-server/rtsp-auth.c:
3810 * gst/rtsp-server/rtsp-auth.h:
3811 * gst/rtsp-server/rtsp-client.h:
3812 auth: add support for multiple basic auth tokens
3813 Make it possible to add multiple basic authorisation tokens to one authorization
3814 object. Associate with each token an authorization group that will define what
3815 capabilities are allowed.
3817 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3819 * gst/rtsp-server/rtsp-client.c:
3820 client: error out on non-aggregate control
3821 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3823 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3825 * gst/rtsp-server/rtsp-client.c:
3826 client: rework setup request a little
3827 Cache the media in DESCRIBE based on the longest matching path with the uri
3828 that we can find in the mount points.
3829 Rework the setup request a little to get the media from the session or from
3830 the longest matching path, this way we can derive the control string as
3831 everything after the path instead of hardcoding it.
3832 Find the stream based on the control string and only open a session when all
3835 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3837 * gst/rtsp-server/rtsp-media.c:
3838 * gst/rtsp-server/rtsp-media.h:
3839 media: add method to find a stream by control url
3841 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3843 * gst/rtsp-server/rtsp-stream.c:
3844 * gst/rtsp-server/rtsp-stream.h:
3845 stream: add method to check control url of stream
3847 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3849 * gst/rtsp-server/rtsp-client.c:
3850 * gst/rtsp-server/rtsp-session-media.c:
3851 * gst/rtsp-server/rtsp-session-media.h:
3852 * gst/rtsp-server/rtsp-session.c:
3853 * gst/rtsp-server/rtsp-session.h:
3854 session: use path matching for session media
3855 Use a path string instead of a uri to lookup session media in the sessions. Also
3856 use path matching to find the largest possible path that matches.
3858 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3860 * gst/rtsp-server/rtsp-client.c:
3861 * gst/rtsp-server/rtsp-mount-points.c:
3862 * gst/rtsp-server/rtsp-mount-points.h:
3863 * tests/check/gst/mountpoints.c:
3864 mount-points: remove useless vmethod
3865 Making lookups in the mount points should not be done with a URL, if there is a
3866 mapping to be done from URL to mount points, we'll need to do it somewhere
3869 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3871 * gst/rtsp-server/rtsp-mount-points.c:
3872 * gst/rtsp-server/rtsp-mount-points.h:
3873 * tests/check/gst/mountpoints.c:
3874 mount-points: improve mount point searching
3875 Use a GSequence to keep track of the mount points.
3876 Match a URL to the longest matching registered mount point. This should be the
3877 URL to perform aggreagate control and the remainder is the stream specific
3879 Add some unit tests for this.
3881 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3883 * gst/rtsp-server/Makefile.am:
3884 rtsp-server: Allow building of static library
3886 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3888 * tests/check/gst/mediafactory.c:
3889 tests: fix compilation
3891 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3893 * gst/rtsp-server/rtsp-sdp.c:
3894 sdp: get control string from stream
3895 Use the control string as configured in the stream.
3897 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3899 * gst/rtsp-server/rtsp-stream.c:
3900 * gst/rtsp-server/rtsp-stream.h:
3901 stream: add methods and property to set control string
3903 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3905 * gst/rtsp-server/rtsp-client.c:
3907 Rename variables for clarity
3908 Keep media in state when we can
3910 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * gst/rtsp-server/rtsp-client.c:
3913 * gst/rtsp-server/rtsp-stream.c:
3914 * gst/rtsp-server/rtsp-stream.h:
3915 stream: add more support for IPv6
3916 Rename _get_address to _get_multicast_address in GstRTSPStream to
3917 make it clear that this function only deals with multicast.
3918 Make it possible to have both an IPv4 and IPv6 multicast address on
3919 a stream. Give the client an IPv4 or IPv6 address depending on the
3920 address it used to connect to the server.
3921 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3923 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3925 * gst/rtsp-server/rtsp-client.c:
3928 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3930 * gst/rtsp-server/rtsp-stream.c:
3931 stream: handle failed port allocation
3932 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3933 can't allocate any family at all. Also keep track of what port families we
3935 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3937 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3939 * gst/rtsp-server/rtsp-stream.c:
3940 stream: improve docs
3942 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3944 * gst/rtsp-server/rtsp-stream-transport.c:
3945 stream-transport: remove old if 0 block
3947 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3949 * tests/check/gst/client.c:
3951 gst_rtsp_client_get_uri() has been removed
3952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3954 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3956 * gst/rtsp-server/rtsp-client.c:
3957 * gst/rtsp-server/rtsp-client.h:
3958 client: add method to filter managed sessions
3959 Add a method to filter the sessions managed by this client connection.
3960 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3962 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3964 * gst/rtsp-server/rtsp-client.c:
3965 * gst/rtsp-server/rtsp-client.h:
3966 client: remove _get_uri() method
3967 Remove the get_uri() method on the client. A client has no uri, the uri
3968 property is an internal property to manage the last cached media for
3971 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3973 * gst/rtsp-server/rtsp-media-factory.h:
3974 media-factory: fix typo
3976 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3978 * gst/rtsp-server/rtsp-media.c:
3979 rtsp-media: Do not leak the query in default_query_stop
3980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3982 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3984 * gst/rtsp-server/rtsp-media.c:
3985 media: don't unlock when conversion fails
3986 Don't unlock the state lock when conversion fails because it was not locked.
3988 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3990 * gst/rtsp-server/rtsp-media.c:
3991 * gst/rtsp-server/rtsp-media.h:
3992 Add query_position and query_stop vmethods to rtsp-media
3994 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3996 * gst/rtsp-server/rtsp-media.c:
3997 Fix typo in property install for rtsp-media's time-provider
3999 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4001 * gst/rtsp-server/rtsp-client.c:
4002 * gst/rtsp-server/rtsp-client.h:
4003 client: clean some variables
4004 Clean some variables and add some guards to _send_request()
4006 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4008 * gst/rtsp-server/rtsp-client.c:
4009 * gst/rtsp-server/rtsp-client.h:
4010 Add gst_rtsp_client_send_request API
4011 This makes it possible to send arbitrary messages to a client, such as
4012 SET_PARAMETER or GET_PARAMETER
4014 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4016 * gst/rtsp-server/rtsp-media.c:
4017 * gst/rtsp-server/rtsp-media.h:
4018 media: add _get_element() method
4019 Add method to get the element used when creating the media.
4020 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4022 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4024 * gst/rtsp-server/rtsp-media.c:
4027 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4029 * gst/rtsp-server/rtsp-stream.c:
4030 * gst/rtsp-server/rtsp-stream.h:
4031 stream: allow access to the rtp session
4032 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4034 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4036 * gst/rtsp-server/rtsp-stream.c:
4037 * gst/rtsp-server/rtsp-stream.h:
4038 dscp qos support in gst-rtsp-stream
4039 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4041 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4043 * tests/check/gst/rtspserver.c:
4045 Actually do what the comment says. Also keep the old code around, not sure what
4046 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4047 it currently doesn't.
4049 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4051 * gst/rtsp-server/rtsp-client.c:
4052 client: also watch newly created session
4053 When we newly created a session, start watching it immediately instead of
4054 on the next request.
4056 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4058 * tests/check/gst/client.c:
4059 tests: add unit test for new-session
4060 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4062 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4064 * gst/rtsp-server/rtsp-client.c:
4065 client: emit new-session when new session is created
4066 Only emit new-session when we created a new session for a client, not when a
4067 client picked up a previous session.
4068 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4070 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4072 * gst/rtsp-server/rtsp-client.c:
4073 client: handle asterisk as path in requests
4074 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4076 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4078 * gst/rtsp-server/rtsp-media.c:
4079 media: handle segment query format mismatch
4080 It's possible that the segment query returns with a different format than what
4081 we asked for, handle this case also.
4083 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4085 * gst/rtsp-server/rtsp-media.c:
4086 media: use segment stop in collect_media_stats
4087 Use segment stop instead of duration as range end point.
4088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4090 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4092 * gst/rtsp-server/rtsp-media.c:
4093 * tests/check/gst/media.c:
4094 rtsp-media: Do not leak the element in take_pipeline
4095 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4097 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4099 * gst/rtsp-server/rtsp-client.c:
4100 * gst/rtsp-server/rtsp-client.h:
4101 rtsp-client: Make configure_client_transport virtual
4102 This patch makes configure_client_transport virtual. The functionality is
4103 needed to handle some weird clients sending multicast transport settings as url
4105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4107 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4109 * gst/rtsp-server/rtsp-client.c:
4110 * gst/rtsp-server/rtsp-client.h:
4111 rtsp-client: Make param_set and param_get virtual
4112 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4114 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4116 * gst/rtsp-server/rtsp-client.c:
4117 * gst/rtsp-server/rtsp-media.c:
4118 * gst/rtsp-server/rtsp-media.h:
4119 media: convert_range replaces get_range_times
4120 get_range_times worked for handling UTC ranges for seeks, but we also
4121 need to convert back from NPT to the requested unit in
4122 get_range_string. convert_range is now used for both.
4123 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4125 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4127 * gst/rtsp-server/rtsp-client.c:
4128 * gst/rtsp-server/rtsp-sdp.c:
4129 * gst/rtsp-server/rtsp-sdp.h:
4130 sdp: cleanup sdp info
4131 We don't need to pass the proto, we can more easily check a boolean.
4132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4134 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4136 * gst/rtsp-server/rtsp-sdp.c:
4137 use 0.0.0.0 or :: for c= line instead of server address
4139 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4141 * gst/rtsp-server/rtsp-client.c:
4142 use local address, not remote, in SDP
4143 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4145 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4148 Automatic update of common submodule
4149 From 098c0d7 to 01a7a46
4151 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4153 * gst/rtsp-server/rtsp-media.c:
4154 * gst/rtsp-server/rtsp-media.h:
4155 media: possibility to override range time conversion
4156 Make it possible to override the conversion from GstRTSPTimeRange to
4157 GstClockTimes, that is done before seeking on the media
4158 pipeline. Overriding can be useful for UTC ranges, where the default
4159 conversion gives nanoseconds since 1900.
4160 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4162 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4164 * gst/rtsp-server/rtsp-server.c:
4165 * gst/rtsp-server/rtsp-server.h:
4166 rtsp-server: Expose the use_client_settings API
4167 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4169 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4171 * gst/rtsp-server/rtsp-client.c:
4172 * gst/rtsp-server/rtsp-stream.c:
4173 * gst/rtsp-server/rtsp-stream.h:
4174 rtspstream: handle both ipv4 and ipv6 clients
4175 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4177 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4179 * gst/rtsp-server/rtsp-sdp.c:
4180 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4181 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4182 We already have a way to place extra attributes in the SDP by using a string
4183 property with prefix x- or a- in the caps.
4185 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4187 * gst/rtsp-server/rtsp-sdp.c:
4188 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4189 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4190 We already have a way to place extra attributes in the SDP, just make a string
4191 property in the payloader with a- or x- prefix.
4193 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4195 * gst/rtsp-server/rtsp-sdp.c:
4196 rtsp: place a- and x- properties as attributes
4197 application/x-rtp has properties with a- and x- prefixes that should be
4198 placed as attributes in the SDP for the media instead of being added to the
4201 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4203 * examples/Makefile.am:
4204 * examples/test-video.c:
4205 example: add TLS example
4207 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4209 * gst/rtsp-server/rtsp-server.c:
4210 * gst/rtsp-server/rtsp-server.h:
4211 server: add support for TLS
4212 Add methods to set and get a TLS certificate.
4213 Add vmethod to configure a new connection. By default, configure the TLS
4214 certificate in a new connection if needed.
4216 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4218 * gst/rtsp-server/rtsp-server.c:
4219 * gst/rtsp-server/rtsp-server.h:
4220 server: remove accept_client vmethod
4221 This vmethod is not very useful so remove it.
4223 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4225 * gst/rtsp-server/rtsp-server.c:
4226 server: don't crash on NULL GError
4228 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4230 * gst/rtsp-server/rtsp-session-pool.c:
4231 rtsp-session-pool: corrected session timeout detection
4232 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4234 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4236 * gst/rtsp-server/rtsp-client.c:
4237 client: improve debug
4239 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4241 * gst/rtsp-server/rtsp-client.c:
4242 * gst/rtsp-server/rtsp-client.h:
4243 * gst/rtsp-server/rtsp-server.c:
4244 server: refactor connection setup
4245 Let the server accept the socket connection and construct a GstRTSPConnection
4246 from it. Remove the code from the client and let the client only deal with
4247 a fully configure GstRTSPConnection object.
4248 We will need this later when the server will configure the connection for
4251 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4253 * gst/rtsp-server/rtsp-stream.c:
4254 stream: keep the transport object alive
4255 Keep the transport object alive while we have it as qdata on the
4258 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4260 * gst/rtsp-server/rtsp-client.c:
4261 * gst/rtsp-server/rtsp-server.c:
4262 rtsp-server: Do not crash on nmapping of server
4263 * generate error when gst_rtsp_connection_accept fails
4264 * do not stop accepting incoming connections because
4265 accepting a client fails
4266 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4268 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4270 * gst/rtsp-server/rtsp-client.c:
4271 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4272 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4274 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4276 * gst/rtsp-server/rtsp-sdp.c:
4277 rtsp-sdp: Parse framerate caps field and set SDP attribute
4278 The SDP attribute and its format is described in RFC4566.
4279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4281 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4283 * gst/rtsp-server/rtsp-sdp.c:
4284 rtsp-sdp: Parse width/height from caps and set SDP attribute
4285 The SDP attribute and its format is described in RFC6064.
4286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4288 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4290 * gst/rtsp-server/rtsp-sdp.c:
4291 * tests/check/gst/client.c:
4292 rtsp-sdp: add bandwidth line
4293 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4295 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4298 Automatic update of common submodule
4299 From 5edcd85 to 098c0d7
4301 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4303 * tests/check/gst/media.c:
4304 tests: add dynamic payloader prepare/unprepare check
4306 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4308 * gst/rtsp-server/rtsp-media.c:
4309 media: release lock when removing fakesink
4311 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4313 * gst/rtsp-server/rtsp-stream.c:
4314 stream: set elements to NULL before removing
4315 When removing a stream, set the elements to NULL first. This avoids
4316 element-is-not-in-NULL-state errors when we dispose the elements.
4318 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4321 Automatic update of common submodule
4322 From 3cb3d3c to 5edcd85
4324 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4326 * gst/rtsp-server/rtsp-media.c:
4327 * gst/rtsp-server/rtsp-media.h:
4328 media: listen to pad-removed signals
4329 Listen to the pad-removed signal and remove the stream associated with the
4331 Add signal to be notified of the removed pad.
4332 Remove the fakesink in unprepare()
4333 Fix signatures of the signal methods
4335 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4337 * examples/test-sdp.c:
4338 tests: add example of reusable pipelines
4340 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4342 * gst/rtsp-server/rtsp-stream.c:
4343 * gst/rtsp-server/rtsp-stream.h:
4344 stream: add method to get the srcpad
4346 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4348 * tests/check/gst/media.c:
4349 check: add media prepare/unprepare test
4350 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4352 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4354 * gst/rtsp-server/rtsp-media.c:
4355 media: disconnect from signal handlers in unprepare()
4356 We connected to the pad-added and no-more-pads signals in prepare() so
4357 we need to disconnect from them in unprepare().
4358 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4360 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4362 * gst/rtsp-server/rtsp-media.c:
4363 media: don't free streams array
4364 Don't free the streams array in the unprepare() method, they were not
4366 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4368 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4370 * gst/rtsp-server/rtsp-media.c:
4371 media: don't unref the pipeline in unprepare
4372 Unprepare() should undo what prepare() does. Because the pipeline is
4373 not created in prepare(), we should not unref it in unprepare()
4375 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4377 * gst/rtsp-server/rtsp-stream.c:
4378 stream: clear session and caps for reuse
4379 Set the session and caps to NULL after unref otherwise we might unref
4381 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4383 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4385 * gst/rtsp-server/rtsp-client.c:
4386 client: send out teardown signal before tearing down
4387 The advantage is that in the signal handler you get direct access to
4388 information about what streams are about to get torn down (in the
4389 GstRTSPClientState).
4390 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4392 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4394 * gst/rtsp-server/rtsp-client.c:
4395 * gst/rtsp-server/rtsp-client.h:
4396 client: expose connection
4397 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4399 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4402 Automatic update of common submodule
4403 From aed87ae to 3cb3d3c
4405 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4407 * gst/rtsp-server/rtsp-media.c:
4408 * gst/rtsp-server/rtsp-media.h:
4409 * gst/rtsp-server/rtsp-session-media.c:
4410 * gst/rtsp-server/rtsp-session-media.h:
4411 media: add method to get the base_time of the pipeline
4412 Together with a shared clock, this base-time could eventually be sent to
4413 the client so that it can reconstruct the exact running-time of the clock
4416 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4418 * gst/rtsp-server/Makefile.am:
4419 * gst/rtsp-server/rtsp-media.c:
4420 * gst/rtsp-server/rtsp-media.h:
4421 * gst/rtsp-server/rtsp-sdp.c:
4422 media: add GstNetTimeProvider support
4423 Add a property to let the media provide a GstNetTimeProvider for its clock.
4424 Make methods to get the clock and nettimeprovider
4425 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4426 provider and also the current time of the clock. This should make it possible
4427 for (GStreamer) clients to slave their clock to the server clock.
4429 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4432 Automatic update of common submodule
4433 From 04c7a1e to aed87ae
4435 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4437 * gst/rtsp-server/rtsp-media.c:
4438 media: wait for buffering to complete
4439 Wait for buffering to complete before changing the state to the target state.
4441 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4443 * gst/rtsp-server/rtsp-media.c:
4444 media: small cleanup
4446 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4448 * tests/check/gst/rtspserver.c:
4449 tests: remove extra unref in test_setup_non_existing_stream
4450 The unref is not needed anymore, teardown runs without it.
4451 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4453 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4455 * tests/check/gst/rtspserver.c:
4456 tests: GSocketService cleanup in test_bind_already_in_use
4457 Use g_socket_service_stop so the rtspserver test stops listening for
4458 incoming connections in test_bind_already_in_use.
4459 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4461 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4463 * gst/rtsp-server/rtsp-media-factory.c:
4464 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4465 Instead use a GWeakRef which is safe to use
4466 This is a known GLib bug, see:
4467 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4469 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4471 * gst/rtsp-server/rtsp-client.c:
4472 * gst/rtsp-server/rtsp-media.c:
4473 * gst/rtsp-server/rtsp-media.h:
4474 * gst/rtsp-server/rtsp-sdp.c:
4475 * tests/check/gst/media.c:
4476 * tests/check/gst/rtspserver.c:
4477 rtsp-media/client: Reply to PLAY request with same type of Range
4478 Remember the type of Range from the PLAY request and use the same type for
4481 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4483 * gst/rtsp-server/rtsp-client.c:
4484 * gst/rtsp-server/rtsp-client.h:
4485 * tests/check/gst/client.c:
4486 rtsp-client: expose uri
4488 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4490 * tests/check/gst/mediafactory.c:
4491 tests: Hold ref while creating second media
4492 To test if the media aren't shared, make sure we keep the first one while creating a second
4493 otherwise the same memory address may be reused.
4495 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4498 configure: remove out-of-date comment
4500 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4503 .gitignore: ignore more build files
4505 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4507 * tests/check/Makefile.am:
4508 tests: use right _LIBS variable for gst-plugins-base libs
4510 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4512 * tests/check/Makefile.am:
4513 check: add librtp to libs
4515 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4517 * tests/check/gst/rtspserver.c:
4518 tests: Add test to check selecting a port the server will send from
4520 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4522 * tests/check/gst/rtspserver.c:
4523 tests: Make sure packets are actually received
4525 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4527 * gst/rtsp-server/rtsp-stream.c:
4528 stream: Select unicast address from pool if appropriate
4530 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4532 * gst/rtsp-server/rtsp-stream.c:
4533 stream: Properties are always there in Gst 1.0
4535 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4537 * tests/check/gst/addresspool.c:
4538 tests: Add tests for unicast addresses in pool
4540 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4542 * gst/rtsp-server/rtsp-address-pool.c:
4543 * tests/check/gst/addresspool.c:
4544 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4546 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4548 * docs/libs/gst-rtsp-server-sections.txt:
4549 * gst/rtsp-server/rtsp-address-pool.c:
4550 * gst/rtsp-server/rtsp-address-pool.h:
4551 * gst/rtsp-server/rtsp-stream.c:
4552 * tests/check/gst/addresspool.c:
4553 address-pool: Add unicast addresses
4555 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4558 * gst/rtsp-server/rtsp-server.c:
4559 * tests/check/gst/rtspserver.c:
4560 rtsp-server: Limit the number of threads per server instance
4561 If we exceed the maximum, just round robin the clients over the existing
4564 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4566 * gst/rtsp-server/rtsp-server.c:
4567 rtsp-server: No need to store the GMainContext in the client context
4569 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4571 * tests/check/gst/rtspserver.c:
4572 tests: Add test for client disconnection
4574 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4576 * tests/check/gst/rtspserver.c:
4577 tests: Test client and session timeouts with multiple threads
4579 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4581 * gst/rtsp-server/rtsp-address-pool.c:
4582 * gst/rtsp-server/rtsp-auth.c:
4583 * gst/rtsp-server/rtsp-client.c:
4584 * gst/rtsp-server/rtsp-media-factory-uri.c:
4585 * gst/rtsp-server/rtsp-media-factory.c:
4586 * gst/rtsp-server/rtsp-media.c:
4587 * gst/rtsp-server/rtsp-mount-points.c:
4588 * gst/rtsp-server/rtsp-server.c:
4589 * gst/rtsp-server/rtsp-session-media.c:
4590 * gst/rtsp-server/rtsp-session-pool.c:
4591 * gst/rtsp-server/rtsp-session.c:
4592 Document locking and its order
4594 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4596 * tests/check/gst/rtspserver.c:
4597 tests: Test that slow DESCRIBE don't block other clients
4599 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4601 * tests/check/gst/client.c:
4602 tests: Add tests for client-requested multicast address
4604 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4606 * docs/libs/gst-rtsp-server-sections.txt:
4607 docs: Put the various functions in the right sections
4609 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4611 * docs/libs/gst-rtsp-server-docs.sgml:
4612 * docs/libs/gst-rtsp-server-sections.txt:
4613 * gst/rtsp-server/rtsp-address-pool.c:
4614 * gst/rtsp-server/rtsp-address-pool.h:
4615 docs: Generate docs for GstRTSPAddressPool
4617 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4619 * gst/rtsp-server/rtsp-client.c:
4620 * gst/rtsp-server/rtsp-stream.c:
4621 * gst/rtsp-server/rtsp-stream.h:
4622 client: Check client provided addresses against the address pool
4624 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4626 * gst/rtsp-server/rtsp-address-pool.c:
4627 * gst/rtsp-server/rtsp-address-pool.h:
4628 * tests/check/gst/addresspool.c:
4629 address-pool: Add API to request a specific address from the pool
4630 Also add relevant unit tests.
4632 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4634 * tests/check/gst/mediafactory.c:
4635 tests: Check the passing around of a RTSPAddressPool
4636 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4637 way down to the stream.
4639 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4641 * tests/check/gst/addresspool.c:
4642 tests: Add more tests for the address pool
4644 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4646 * gst/rtsp-server/rtsp-address-pool.c:
4647 address-pool: Fix off by one error
4648 When splitting a port range, the port after a skip is not part of range.
4650 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4653 Automatic update of common submodule
4654 From 2de221c to 04c7a1e
4656 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4659 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4660 AM_CONFIG_HEADER was removed in automake 1.13
4661 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4663 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4666 Automatic update of common submodule
4667 From a942293 to 2de221c
4669 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4671 * gst/rtsp-server/rtsp-client.c:
4672 client: make sure the watch exists while sending data
4673 Protect the send_func with a lock. This allows us to wait for sending
4674 to complete before changing the send_func and user_data. We add an
4675 extra ref to the watch to make sure that it remains valid during
4677 When closing the connection, set the send_func to NULL
4678 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4680 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4682 * tests/check/Makefile.am:
4683 tests: use GST_*_1_0 environment variables everywhere
4684 The _1_0 suffixed environment variables override the
4685 non-suffixed ones, so if we're in an environment that
4686 sets the _1_0 suffixed ones, such as jhbuild, we need
4687 to set those to make sure ours actually always get
4690 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4693 Automatic update of common submodule
4694 From acb04d9 to a942293
4696 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4698 * gst/rtsp-server/rtsp-client.c:
4699 rtsp-client: set the client backlog
4700 Set the client backlog to a reasonable default
4702 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4704 * gst/rtsp-server/rtsp-media.c:
4705 rtsp-media: Make the element a constructor parameter
4706 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4708 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4710 * docs/libs/Makefile.am:
4711 docs: Link with gcov library when gcov is enabled
4712 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4714 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * gst/rtsp-server/rtsp-media.c:
4717 media: match prepare with unprepare
4718 Really unprepare when there were an equal amount of prepare calls.
4720 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4722 * gst/rtsp-server/rtsp-media.c:
4723 media: media has to be unprepared in finalize
4724 Because unprepare takes away the last ref on the media.
4726 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-client.c:
4729 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4730 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4731 We can't use the refcount to trigger unprepare because it is the unprepare call
4732 that removes the last refcount after all messages are consumed. What we should
4733 probably do is make a prepared refcount and only unprepare when the refcount
4736 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4738 * gst/rtsp-server/rtsp-media.c:
4739 media: let the source unref the last media ref
4740 the last ref to the media is held by the source so we don't need to add more ref
4741 and unrefs, we simply destroy the media when the source is gone.
4743 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4745 * gst/rtsp-server/rtsp-media.c:
4746 media: improve debug
4748 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4750 * gst/rtsp-server/rtsp-media.c:
4752 Make sure we are in the right state when collecting the position and duration.
4753 Only make ourselves PREPARED when we were previously PREPARING.
4755 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4757 * gst/rtsp-server/rtsp-media.c:
4758 media: use g_object_ref/unref for GObjects
4760 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4762 * gst/rtsp-server/rtsp-client.c:
4763 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4764 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4765 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4766 isn't being used anymore.
4768 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4770 * gst/rtsp-server/rtsp-media.c:
4771 Fix compiler warning
4773 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4775 * gst/rtsp-server/rtsp-media-factory-uri.c:
4776 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4778 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4780 * gst/rtsp-server/rtsp-session-media.h:
4783 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * gst/rtsp-server/rtsp-media.c:
4786 * tests/check/gst/media.c:
4787 media: avoid element leak
4789 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4791 * gst/rtsp-server/rtsp-media.c:
4792 media: require an element in media constructor
4794 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4796 * gst/rtsp-server/rtsp-client.c:
4797 Revert "client: TEARDOWN brings that state to Init again"
4798 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4799 The object is already disposed, there is no point in setting the state.
4801 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * gst/rtsp-server/rtsp-client.c:
4804 client: TEARDOWN brings that state to Init again
4806 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4808 * docs/libs/gst-rtsp-server-sections.txt:
4809 * examples/test-auth.c:
4810 * gst/rtsp-server/rtsp-auth.c:
4811 * gst/rtsp-server/rtsp-auth.h:
4812 * gst/rtsp-server/rtsp-client.c:
4813 * gst/rtsp-server/rtsp-client.h:
4814 * gst/rtsp-server/rtsp-media-factory-uri.c:
4815 * gst/rtsp-server/rtsp-media-factory-uri.h:
4816 * gst/rtsp-server/rtsp-media-factory.c:
4817 * gst/rtsp-server/rtsp-media-factory.h:
4818 * gst/rtsp-server/rtsp-media.c:
4819 * gst/rtsp-server/rtsp-media.h:
4820 * gst/rtsp-server/rtsp-mount-points.c:
4821 * gst/rtsp-server/rtsp-mount-points.h:
4822 * gst/rtsp-server/rtsp-sdp.c:
4823 * gst/rtsp-server/rtsp-server.c:
4824 * gst/rtsp-server/rtsp-server.h:
4825 * gst/rtsp-server/rtsp-session-media.c:
4826 * gst/rtsp-server/rtsp-session-media.h:
4827 * gst/rtsp-server/rtsp-session-pool.c:
4828 * gst/rtsp-server/rtsp-session-pool.h:
4829 * gst/rtsp-server/rtsp-session.c:
4830 * gst/rtsp-server/rtsp-session.h:
4831 * gst/rtsp-server/rtsp-stream-transport.c:
4832 * gst/rtsp-server/rtsp-stream-transport.h:
4833 * gst/rtsp-server/rtsp-stream.c:
4834 * gst/rtsp-server/rtsp-stream.h:
4835 * tests/check/gst/media.c:
4836 rtsp: make object details private
4837 Make all object details private
4838 Add methods to access private bits
4840 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4842 * tests/check/Makefile.am:
4843 * tests/check/gst/media.c:
4844 tests: add media tests
4846 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4848 * gst/rtsp-server/rtsp-media.c:
4849 media: check if prepared for some methods
4850 Check that the media object is prepared before doing seek and getting the
4851 current position etc.
4852 Add some g_return checks.
4854 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4856 * tests/check/Makefile.am:
4857 * tests/check/gst/mediafactory.c:
4858 tests: add mediafactory test
4860 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4862 * gst/rtsp-server/rtsp-stream.c:
4863 stream: improve debug
4865 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4867 * gst/rtsp-server/rtsp-media.c:
4868 * gst/rtsp-server/rtsp-media.h:
4869 media: unref pipeline in finalize to avoid leaking it
4871 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4873 * gst/rtsp-server/rtsp-media-factory-uri.c:
4874 * gst/rtsp-server/rtsp-media.c:
4875 rtsp: use gst_object_unref on GstObjects
4877 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4879 * gst/rtsp-server/rtsp-media-factory.c:
4880 media-factory: require an url
4882 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4884 * examples/test-uri.c:
4885 examples: fix include
4887 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4889 * gst/rtsp-server/rtsp-server.h:
4890 server: remove unused include
4892 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4894 * tests/check/Makefile.am:
4895 * tests/check/gst/mountpoints.c:
4896 tests: add test for mountpoints
4898 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4900 * gst/rtsp-server/rtsp-client.c:
4901 client: fix factory leak
4902 Keep the factory in the state object only for authorization checks and make
4903 sure we unref it on failure. Also don't keep invalid objects in the state
4906 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4908 * gst/rtsp-server/rtsp-mount-points.c:
4909 mounts: add g_return_if guards
4911 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4913 * tests/check/gst/client.c:
4914 tests: add more tests
4916 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4918 * gst/rtsp-server/rtsp-client.c:
4919 client: improve debug
4921 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4923 * gst/rtsp-server/rtsp-client.c:
4924 client: improve debug and fix leaks
4925 Cleanup the uri and session when there is a bad request.
4927 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4932 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4934 * tests/check/gst/client.c:
4935 test: add test for session in options request
4937 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4939 * gst/rtsp-server/rtsp-client.c:
4940 client: use 454 when session can't be found
4941 We should use 454 when a session can't be found because there was no session
4942 pool configured in the server. This is not a server configuration problem
4943 because the server on which the request is done might not be the same one that
4944 will keep the sessions for us and so it does not need to support sessions.
4946 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4948 * gst/rtsp-server/rtsp-client.c:
4949 client: only free connection when there is one
4950 It's possible that the client doesn't have a connection when we try to free it.
4952 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4954 * tests/check/Makefile.am:
4955 * tests/check/gst/client.c:
4956 tests: add unit test for the client object
4958 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4960 * gst/rtsp-server/rtsp-client.c:
4961 client: small cleanup
4963 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4965 * gst/rtsp-server/rtsp-client.h:
4966 client: remove unused include
4968 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4970 * gst/rtsp-server/rtsp-client.c:
4971 client: fix compilation
4973 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4975 * gst/rtsp-server/rtsp-client.c:
4976 client: call destroy without the lock
4978 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4980 * gst/rtsp-server/rtsp-client.c:
4981 * gst/rtsp-server/rtsp-client.h:
4982 client: make the client usable without a socket
4983 Make a method to let the client handle a message and a callback when the client
4984 wants us to send a response message back. This makes it possible to also use the
4985 client object without the sockets, which should make it easier to test.
4987 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4989 * gst/rtsp-server/rtsp-client.c:
4990 * gst/rtsp-server/rtsp-client.h:
4991 client: small cleanup
4993 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4995 * docs/libs/gst-rtsp-server-sections.txt:
4996 * gst/rtsp-server/rtsp-client.c:
4997 * gst/rtsp-server/rtsp-client.h:
4998 * gst/rtsp-server/rtsp-server.c:
4999 client: remove reference to server
5000 We don't need to keep a ref to the server
5002 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5004 * gst/rtsp-server/rtsp-client.c:
5005 * gst/rtsp-server/rtsp-client.h:
5007 Also add some g_return_if()
5009 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5011 * gst/rtsp-server/rtsp-client.c:
5012 client: log more errors
5014 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5016 * gst/rtsp-server/rtsp-client.c:
5017 client: fix compilation
5019 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5021 * gst/rtsp-server/rtsp-client.c:
5022 * gst/rtsp-server/rtsp-client.h:
5023 client: add generic close-after-send support
5024 Add a property to send_response() to close the connection after the response has
5025 been sent to the client.
5027 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5030 * docs/libs/gst-rtsp-server-docs.sgml:
5031 * docs/libs/gst-rtsp-server-sections.txt:
5032 * docs/libs/gst-rtsp-server.types:
5033 * examples/test-auth.c:
5034 * examples/test-launch.c:
5035 * examples/test-mp4.c:
5036 * examples/test-multicast.c:
5037 * examples/test-multicast2.c:
5038 * examples/test-ogg.c:
5039 * examples/test-readme.c:
5040 * examples/test-sdp.c:
5041 * examples/test-uri.c:
5042 * examples/test-video.c:
5043 * gst/rtsp-server/Makefile.am:
5044 * gst/rtsp-server/rtsp-auth.h:
5045 * gst/rtsp-server/rtsp-client.c:
5046 * gst/rtsp-server/rtsp-client.h:
5047 * gst/rtsp-server/rtsp-media-mapping.c:
5048 * gst/rtsp-server/rtsp-media-mapping.h:
5049 * gst/rtsp-server/rtsp-mount-points.c:
5050 * gst/rtsp-server/rtsp-mount-points.h:
5051 * gst/rtsp-server/rtsp-server.c:
5052 * gst/rtsp-server/rtsp-server.h:
5053 * gst/rtsp-server/rtsp-session-media.c:
5054 * gst/rtsp-server/rtsp-session-pool.c:
5055 * gst/rtsp-server/rtsp-session-pool.h:
5056 * tests/check/gst/rtspserver.c:
5057 MediaMapping -> MountPoints
5058 Describes better what the object manages.
5060 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5063 configure: bump required version of -base
5065 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5067 * gst/rtsp-server/rtsp-media.c:
5070 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5072 * gst/rtsp-server/rtsp-media.c:
5073 * gst/rtsp-server/rtsp-media.h:
5074 media: support more Range formats
5075 Use the new -base methods to convert the Range string into a seek start and stop
5078 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * examples/test-launch.c:
5081 examples: fix whitespace
5083 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5085 * examples/test-auth.c:
5086 test-auth: add example of how to remove sessions
5087 Add an example of the session filter api.
5089 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5091 * examples/test-uri.c:
5092 test-uri: remove mapping example
5094 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5096 * examples/test-uri.c:
5097 test-uri: fix callback signature
5099 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5101 * gst/rtsp-server/rtsp-media-factory.c:
5102 factory: keep ref to factory while media active
5103 While the media from a factory is alive, keep a ref to the factory.
5104 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5106 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5108 * gst/rtsp-server/rtsp-media-factory-uri.c:
5109 factory-uri: add some debug
5111 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5113 * gst/rtsp-server/rtsp-stream.c:
5114 stream: set udp sources to PLAYING
5115 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5116 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5118 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5120 * gst/rtsp-server/rtsp-media-factory-uri.c:
5121 factory-uri: take ref to factory
5122 Take a ref to the factory that we place in our list.
5124 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5126 * tests/Makefile.am:
5127 * tests/test-reuse.c:
5128 test: add test for server reuse
5129 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5131 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5133 * gst/rtsp-server/rtsp-server.c:
5134 server: start and stop multiple times
5135 Stop listening on the RTSP port when the GSource is removed, so clients
5136 can't connect and the server can be started again.
5137 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5139 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5141 * gst/rtsp-server/rtsp-server.c:
5142 server: fix small leak
5144 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5146 * gst/rtsp-server/rtsp-media.c:
5147 media: unref source in finish_unprepare
5148 The source is created in prepare, unref it in finish_unprepare.
5149 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5151 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5153 * gst/rtsp-server/rtsp-client.c:
5154 * gst/rtsp-server/rtsp-media.c:
5155 rtsp-media: remove bus watch before finalizing
5156 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5157 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5158 the GDestroyNotify function.
5159 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5160 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5161 gst_rtsp_media_unprepare before unreffing the media.
5162 This way, the bus watch will be removed before the media is finalized.
5163 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5165 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5167 * gst/rtsp-server/rtsp-client.c:
5168 * gst/rtsp-server/rtsp-client.h:
5169 client: wait until the TEARDOWN response is sent to close the connection
5170 Responses can be sent async so we need to wait until the TEARDOWN response has
5171 been written before we close the connection to the client. This avoids the risk
5172 of writing/polling closed sockets.
5173 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5175 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5177 * gst/rtsp-server/rtsp-stream.c:
5178 rtsp-stream: plug socket leak
5179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5181 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5184 Automatic update of common submodule
5185 From 6bb6951 to a72faea
5187 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5189 * gst/rtsp-server/rtsp-media-factory-uri.c:
5190 rtsp-server: don't use deprecated API
5192 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5194 * gst/rtsp-server/rtsp-client.c:
5195 rtsp-client: fix unused-but-set-variable compiler warning
5196 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5198 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5201 * docs/libs/gst-rtsp-server-sections.txt:
5202 * gst/rtsp-server/rtsp-client.c:
5205 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5207 * examples/Makefile.am:
5208 * examples/test-multicast2.c:
5209 examples: add another multicast example
5210 Add an example for how to configure separate multicast ranges for each media
5213 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5215 * examples/test-multicast.c:
5218 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5220 * gst/rtsp-server/rtsp-client.c:
5221 * gst/rtsp-server/rtsp-media.c:
5222 * gst/rtsp-server/rtsp-session-media.c:
5223 * gst/rtsp-server/rtsp-session-media.h:
5224 * gst/rtsp-server/rtsp-stream-transport.c:
5225 * gst/rtsp-server/rtsp-stream-transport.h:
5226 stream: use the address managed by the stream
5227 Use the address managed by the stream for multicast. This allows us to have 1
5228 multicast address for each stream.
5229 Because the address is now managed by the stream we don't have to pass it around
5231 Set the address pool on the streams.
5233 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5235 * gst/rtsp-server/rtsp-client.c:
5236 * gst/rtsp-server/rtsp-media.c:
5237 * gst/rtsp-server/rtsp-stream.c:
5240 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5242 * gst/rtsp-server/rtsp-media.c:
5243 * gst/rtsp-server/rtsp-media.h:
5244 media: add signal for new streams
5245 This allows applications to listen for new streams and configure properties on
5246 them, like the address pool.
5248 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5250 * gst/rtsp-server/rtsp-media.c:
5251 media: configure address pool in new streams
5253 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5255 * gst/rtsp-server/rtsp-stream.c:
5256 * gst/rtsp-server/rtsp-stream.h:
5257 stream: add methods to deal with address pool
5258 Add methods to get and set the address pool for the stream
5259 Add method to allocate and get the multicast addresses for this stream.
5261 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5263 * docs/libs/gst-rtsp-server-sections.txt:
5264 * gst/rtsp-server/rtsp-media.c:
5265 * gst/rtsp-server/rtsp-media.h:
5266 media: remove MTU property
5267 It is a stream property
5269 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5271 * gst/rtsp-server/rtsp-client.c:
5272 client: set blocksize only on stream
5273 Set the blocksize only on the current stream.
5275 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5277 * gst/rtsp-server/rtsp-stream.c:
5278 stream: share src and sink sockets
5279 the allocated socket is in the used-socket property, not socket.
5281 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5283 * gst/rtsp-server/rtsp-address-pool.c:
5284 * gst/rtsp-server/rtsp-address-pool.h:
5285 * gst/rtsp-server/rtsp-client.c:
5286 * gst/rtsp-server/rtsp-session-media.c:
5287 * gst/rtsp-server/rtsp-session-media.h:
5288 * gst/rtsp-server/rtsp-stream-transport.c:
5289 * gst/rtsp-server/rtsp-stream-transport.h:
5290 * tests/check/gst/addresspool.c:
5291 rtsp: make address-pool return an address object
5292 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5293 store more info in the structure and allows us to more easily return the address
5294 to the right pool when no longer needed.
5295 Pass the address to the StreamTransport so that we can return it to the pool
5296 when the stream transport is freed or changed.
5298 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5300 * examples/Makefile.am:
5301 * examples/test-multicast.c:
5302 examples: add multicast example
5303 Show how to set up the multicast address pool so that media can be
5304 server with multicast.
5306 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5308 * gst/rtsp-server/rtsp-client.c:
5309 * gst/rtsp-server/rtsp-media-factory.c:
5310 * gst/rtsp-server/rtsp-media-factory.h:
5311 * gst/rtsp-server/rtsp-media.c:
5312 * gst/rtsp-server/rtsp-media.h:
5313 rtsp: use AddressPool
5314 Remove the multicast_group property.
5315 Use the configured addresspool to allocate multicast addresses.
5317 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5319 * gst/rtsp-server/rtsp-address-pool.c:
5320 * gst/rtsp-server/rtsp-address-pool.h:
5321 address-pool: add clear method
5323 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5325 * gst/rtsp-server/rtsp-address-pool.c:
5326 address-pool: small cleanups
5328 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5330 * tests/check/Makefile.am:
5331 * tests/check/gst/addresspool.c:
5332 tests: add addresspool unit test
5334 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5336 * gst/rtsp-server/Makefile.am:
5337 * gst/rtsp-server/rtsp-address-pool.c:
5338 * gst/rtsp-server/rtsp-address-pool.h:
5339 address-pool: add object to manage multicast addresses
5340 Make an object that can manage a rage of multicast addresses and ports.
5342 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5344 * gst/rtsp-server/rtsp-server.c:
5345 server: set default max-threads property
5347 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5349 * gst/rtsp-server/rtsp-media.c:
5350 media: wait for concurrent _prepare
5351 If a prepare is busy, wait for the result.
5353 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5355 * gst/rtsp-server/rtsp-media.c:
5356 media: add lock around message handler
5357 We don't want to dispatch messages while we are still processing the result of
5360 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5362 * gst/rtsp-server/rtsp-media.c:
5363 * gst/rtsp-server/rtsp-media.h:
5364 media: add lock to protect state changes
5366 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5368 * gst/rtsp-server/rtsp-stream.c:
5369 * gst/rtsp-server/rtsp-stream.h:
5372 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5374 * gst/rtsp-server/rtsp-stream-transport.c:
5375 * gst/rtsp-server/rtsp-stream-transport.h:
5376 * gst/rtsp-server/rtsp-stream.c:
5377 stream-transport: add keep-alive method
5379 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5381 * gst/rtsp-server/rtsp-stream-transport.c:
5382 * gst/rtsp-server/rtsp-stream-transport.h:
5383 * gst/rtsp-server/rtsp-stream.c:
5384 stream-transport: add method to handle RTP/RTCP
5385 Call new methods instead of poking into the structures directly.
5387 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5389 * gst/rtsp-server/rtsp-session-media.c:
5390 * gst/rtsp-server/rtsp-session-media.h:
5391 session-media: add locking
5393 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5395 * gst/rtsp-server/rtsp-session.c:
5396 * gst/rtsp-server/rtsp-session.h:
5397 session: add locking
5399 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5401 * gst/rtsp-server/rtsp-server.c:
5402 server: free old socket
5404 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5406 * gst/rtsp-server/rtsp-media-mapping.c:
5407 * gst/rtsp-server/rtsp-media-mapping.h:
5408 mapping: add locking
5410 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5412 * gst/rtsp-server/rtsp-media-factory.c:
5413 media-factory: add locking
5415 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5417 * gst/rtsp-server/rtsp-auth.c:
5418 * gst/rtsp-server/rtsp-auth.h:
5421 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5423 * gst/rtsp-server/rtsp-server.c:
5424 * gst/rtsp-server/rtsp-server.h:
5425 server: add max-thread property
5427 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5429 * gst/rtsp-server/rtsp-server.c:
5430 * gst/rtsp-server/rtsp-server.h:
5431 server: use a threadpool for the mainloops
5433 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5435 * gst/rtsp-server/rtsp-client.c:
5436 * gst/rtsp-server/rtsp-client.h:
5437 client: rename method
5438 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5439 don't really create the client from the socket, we use the socket for the
5442 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5444 * gst/rtsp-server/rtsp-client.c:
5445 * gst/rtsp-server/rtsp-client.h:
5446 * gst/rtsp-server/rtsp-server.c:
5447 server: rework maincontext handling in clients
5448 Make a separate method to attach a client to a MainContext.
5449 Let the server decide in what GMainContext the client will operate and give this
5450 context to the client in attach. Then the server can later decide to use a
5451 separate thread for each client or just use the mainthread.
5453 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5455 * gst/rtsp-server/rtsp-client.c:
5456 * gst/rtsp-server/rtsp-session.c:
5457 * gst/rtsp-server/rtsp-session.h:
5458 session: move session header code in session object
5460 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5464 * examples/test-auth.c:
5465 * examples/test-launch.c:
5466 * examples/test-mp4.c:
5467 * examples/test-ogg.c:
5468 * examples/test-readme.c:
5469 * examples/test-sdp.c:
5470 * examples/test-uri.c:
5471 * examples/test-video.c:
5472 * gst/rtsp-server/rtsp-auth.c:
5473 * gst/rtsp-server/rtsp-auth.h:
5474 * gst/rtsp-server/rtsp-client.c:
5475 * gst/rtsp-server/rtsp-client.h:
5476 * gst/rtsp-server/rtsp-media-factory-uri.c:
5477 * gst/rtsp-server/rtsp-media-factory-uri.h:
5478 * gst/rtsp-server/rtsp-media-factory.c:
5479 * gst/rtsp-server/rtsp-media-factory.h:
5480 * gst/rtsp-server/rtsp-media-mapping.c:
5481 * gst/rtsp-server/rtsp-media-mapping.h:
5482 * gst/rtsp-server/rtsp-media.c:
5483 * gst/rtsp-server/rtsp-media.h:
5484 * gst/rtsp-server/rtsp-params.c:
5485 * gst/rtsp-server/rtsp-params.h:
5486 * gst/rtsp-server/rtsp-sdp.c:
5487 * gst/rtsp-server/rtsp-sdp.h:
5488 * gst/rtsp-server/rtsp-server.c:
5489 * gst/rtsp-server/rtsp-server.h:
5490 * gst/rtsp-server/rtsp-session-media.c:
5491 * gst/rtsp-server/rtsp-session-media.h:
5492 * gst/rtsp-server/rtsp-session-pool.c:
5493 * gst/rtsp-server/rtsp-session-pool.h:
5494 * gst/rtsp-server/rtsp-session.c:
5495 * gst/rtsp-server/rtsp-session.h:
5496 * gst/rtsp-server/rtsp-stream-transport.c:
5497 * gst/rtsp-server/rtsp-stream-transport.h:
5498 * gst/rtsp-server/rtsp-stream.c:
5499 * gst/rtsp-server/rtsp-stream.h:
5500 * tests/check/gst/rtspserver.c:
5501 * tests/test-cleanup.c:
5504 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5506 * gst/rtsp-server/rtsp-media.c:
5507 * gst/rtsp-server/rtsp-session-media.c:
5508 * gst/rtsp-server/rtsp-session.c:
5509 rtsp-server: added annotations to indicate type of ownership transfer of return values
5510 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5512 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5515 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5517 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5520 * bindings/Makefile.am:
5521 * bindings/vala/Makefile.am:
5522 * bindings/vala/gst-rtsp-server-0.10.deps:
5523 * bindings/vala/gst-rtsp-server-0.10.vapi:
5524 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5525 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5526 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5527 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5528 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5530 bindings: remove vala bindings
5531 They'll be reunited with the other GStreamer bindings
5532 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5534 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5536 * gst/rtsp-server/rtsp-client.c:
5537 * gst/rtsp-server/rtsp-session-media.c:
5538 * gst/rtsp-server/rtsp-session-media.h:
5539 * gst/rtsp-server/rtsp-stream-transport.c:
5540 * gst/rtsp-server/rtsp-stream-transport.h:
5541 rtsp: only create transport when needed
5542 Only create the StreamTransport when configured.
5544 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5546 * gst/rtsp-server/rtsp-client.c:
5547 client: small cleanup
5549 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5551 * gst/rtsp-server/rtsp-client.c:
5552 * gst/rtsp-server/rtsp-client.h:
5553 * gst/rtsp-server/rtsp-stream-transport.c:
5554 * gst/rtsp-server/rtsp-stream-transport.h:
5555 rtsp: refactor configuration of transport
5556 Move the configuration of the transport to a place where it makes
5559 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5561 * gst/rtsp-server/rtsp-client.c:
5562 client: refactor transport parsing
5564 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5566 * gst/rtsp-server/rtsp-client.c:
5567 client: refuse to change the MTU on shared media
5568 If we change the MTU of chared media, it changes for all clients.
5569 We don't want to set the MTU to something large for clients that
5572 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5574 * examples/test-mp4.c:
5575 * gst/rtsp-server/rtsp-media.c:
5576 small fixes to docs and debug
5578 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5580 * gst/rtsp-server/rtsp-stream.c:
5581 stream: transports must already have been removed
5583 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5585 * gst/rtsp-server/rtsp-media.c:
5586 * gst/rtsp-server/rtsp-stream.c:
5587 * gst/rtsp-server/rtsp-stream.h:
5588 stream: improve join and leave of the pipeline
5590 Do the cleanup properly
5593 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5595 * gst/rtsp-server/rtsp-media.c:
5596 media: move unprepare below default implementation
5597 Makes it easier to find the default implementation
5599 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5601 * gst/rtsp-server/rtsp-media.c:
5602 media: signal unprepared when we actually finish
5604 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5606 * gst/rtsp-server/rtsp-media.c:
5607 media: no need to unlock, unprepare does that when needed
5609 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5611 * docs/libs/gst-rtsp-server-sections.txt:
5612 * gst/rtsp-server/rtsp-media-factory.h:
5613 * gst/rtsp-server/rtsp-media-mapping.c:
5614 * gst/rtsp-server/rtsp-media.h:
5615 * gst/rtsp-server/rtsp-params.c:
5616 * gst/rtsp-server/rtsp-server.c:
5617 * gst/rtsp-server/rtsp-session-pool.h:
5618 * gst/rtsp-server/rtsp-session.c:
5619 * gst/rtsp-server/rtsp-session.h:
5620 * gst/rtsp-server/rtsp-stream-transport.h:
5621 * gst/rtsp-server/rtsp-stream.h:
5624 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5626 * gst/rtsp-server/rtsp-client.c:
5627 * gst/rtsp-server/rtsp-media-mapping.h:
5628 * gst/rtsp-server/rtsp-media.c:
5629 * gst/rtsp-server/rtsp-media.h:
5630 * gst/rtsp-server/rtsp-server.h:
5631 * gst/rtsp-server/rtsp-stream.c:
5632 * gst/rtsp-server/rtsp-stream.h:
5633 rtsp: fix MTU setting
5634 Fix setting of the MTU. There is no need for a vmethod.
5636 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5641 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5644 configure: bump version number after refactoring
5646 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5648 * gst/rtsp-server/Makefile.am:
5649 * gst/rtsp-server/rtsp-client.c:
5650 * gst/rtsp-server/rtsp-client.h:
5651 * gst/rtsp-server/rtsp-media-factory-uri.c:
5652 * gst/rtsp-server/rtsp-media-factory.c:
5653 * gst/rtsp-server/rtsp-media-factory.h:
5654 * gst/rtsp-server/rtsp-media.c:
5655 * gst/rtsp-server/rtsp-media.h:
5656 * gst/rtsp-server/rtsp-sdp.c:
5657 * gst/rtsp-server/rtsp-session-media.c:
5658 * gst/rtsp-server/rtsp-session-media.h:
5659 * gst/rtsp-server/rtsp-session.c:
5660 * gst/rtsp-server/rtsp-session.h:
5661 * gst/rtsp-server/rtsp-stream-transport.c:
5662 * gst/rtsp-server/rtsp-stream-transport.h:
5663 * gst/rtsp-server/rtsp-stream.c:
5664 * gst/rtsp-server/rtsp-stream.h:
5665 rtsp: massive refactoring
5666 Make GObjects from the remaining simple structures.
5667 Remove GstRTSPSessionStream, it's not needed.
5668 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5669 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5670 a GstRTSPStream should be transported to a client.
5671 Rename GstRTSPMediaFactory::get_element -> create_element because that
5672 more accurately describes what it does.
5673 Make nice methods instead of poking in the structures.
5674 Move some methods inside the relevant object source code.
5675 Use GPtrArray to store objects instead of plain arrays, it is more
5676 natural and allows us to more easily clean up.
5677 Move the allocation of udp ports to the Stream object. The Stream object
5678 contains the elements needed to stream the media to a client.
5679 Improve the prepare and unprepare methods. Unprepare should now undo
5680 everything prepare did. Improve also async unprepare when doing EOS on
5681 shutdown. Make sure we always unprepare correctly.
5683 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5685 * gst/rtsp-server/rtsp-client.c:
5686 rtsp-client: Unref server address clients connected to
5687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5689 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5691 * gst/rtsp-server/rtsp-server.c:
5692 rtsp-server: don't ref server socket if it is NULL
5693 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5694 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5696 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5698 * tests/check/Makefile.am:
5699 tests: Add libgio link dependency
5700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5702 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5704 * gst/rtsp-server/rtsp-media-mapping.c:
5705 * gst/rtsp-server/rtsp-media-mapping.h:
5706 rtsp-media-mapping: rename find_media vfunc to find_factory
5707 The virtual method and class method should have the same name
5708 so it is correctly represented in GIR file
5709 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5711 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5713 * gst/rtsp-server/rtsp-auth.c:
5714 * gst/rtsp-server/rtsp-client.c:
5715 * gst/rtsp-server/rtsp-media-factory-uri.c:
5716 * gst/rtsp-server/rtsp-media-factory.c:
5717 * gst/rtsp-server/rtsp-media-mapping.c:
5718 * gst/rtsp-server/rtsp-media.c:
5719 * gst/rtsp-server/rtsp-server.c:
5720 * gst/rtsp-server/rtsp-session-pool.c:
5721 * gst/rtsp-server/rtsp-session.c:
5722 rtsp-server: fixed comments and GIR annotations
5723 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5725 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5727 * gst/rtsp-server/rtsp-media-mapping.c:
5728 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5730 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5732 * gst/rtsp-server/rtsp-server.c:
5733 rtsp-server: allow binding on port 0 (binds on a random port)
5735 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5737 * gst/rtsp-server/rtsp-server.c:
5738 * gst/rtsp-server/rtsp-server.h:
5739 rtsp-server: add bound-port property
5740 bound-port can be used to retrieve the port number when the server is bound on
5741 port 0, which binds on a random port.
5743 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5745 * gst/rtsp-server/rtsp-media-factory.c:
5746 * gst/rtsp-server/rtsp-media-factory.h:
5747 rtsp-media-factory: make ::get_element overridable by GI bindings
5748 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5749 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5750 as the invoker for ::get_element(), making it overridable by GI generated
5753 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5755 * gst/rtsp-server/rtsp-media-factory-uri.c:
5756 rtsp-media-factory-uri: don't autoplug parsers in a loop
5757 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5760 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5762 * gst/rtsp-server/Makefile.am:
5763 Explicitly link against gio. Fix link error on mac.
5765 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5767 * gst/rtsp-server/rtsp-session.c:
5768 session: add ttl to the transport header in SETUP
5769 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5771 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5773 * gst/rtsp-server/rtsp-client.c:
5774 * gst/rtsp-server/rtsp-client.h:
5775 * gst/rtsp-server/rtsp-media.c:
5776 client: Use client transport settings for multicast if allowed.
5777 This patch makes it possible for the client to send transport settings for
5778 multicast (destination && ttl). Client settings must be explicitly allowed or
5779 the server will use its own settings.
5780 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5782 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5785 Automatic update of common submodule
5786 From 6c0b52c to 6bb6951
5788 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5790 * gst/rtsp-server/rtsp-client.c:
5791 rtsp-client: do not destroy the rtsp watch
5792 Don't destroy the client watch while dispatching. The rtsp watch is
5793 automatically destroyed after the rtsp watch function closed() has
5795 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5797 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5800 Automatic update of common submodule
5801 From 4f962f7 to 6c0b52c
5803 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5805 * gst/rtsp-server/rtsp-media.c:
5806 media: fix check for seekability
5808 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5810 * gst/rtsp-server/rtsp-client.c:
5811 client: use more GIO
5812 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5814 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5816 * gst/rtsp-server/rtsp-server.c:
5817 server: remove obsolete includes
5819 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5821 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5822 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5823 be available in "on_new_ssrc". The transports are added in
5824 gst_rtsp_media_set_state when going to PLAYING state. However,
5825 "on_new_ssrc" might be called before this happens.
5826 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5828 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5830 * gst/rtsp-server/rtsp-client.c:
5831 * gst/rtsp-server/rtsp-client.h:
5832 rtsp-client: add signals for rtsp requests (fixes #683287)
5834 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5836 * gst/rtsp-server/rtsp-client.c:
5837 * gst/rtsp-server/rtsp-client.h:
5838 add new-session signal to rtsp-client (fixes #683058)
5840 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5843 Automatic update of common submodule
5844 From 668acee to 4f962f7
5846 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5848 * gst/rtsp-server/rtsp-server.c:
5849 * tests/check/gst/rtspserver.c:
5850 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5851 Do not assume that *error is set in g_socket_address_enumerator_next.
5852 Added test_bind_already_in_use unit-test.
5853 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5855 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5858 Automatic update of common submodule
5859 From 94ccf4c to 668acee
5861 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5863 * gst/rtsp-server/rtsp-client.c:
5864 * gst/rtsp-server/rtsp-client.h:
5865 rtsp-client: make create_sdp virtual method
5866 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5868 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5871 Automatic update of common submodule
5872 From 98e386f to 94ccf4c
5874 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5876 * gst/rtsp-server/rtsp-client.c:
5879 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5881 * gst/rtsp-server/rtsp-client.c:
5882 * gst/rtsp-server/rtsp-client.h:
5883 * gst/rtsp-server/rtsp-server.c:
5884 * gst/rtsp-server/rtsp-server.h:
5885 rtsp-server: use an existing socket to establish HTTP tunnel
5886 Make it possible to transfer a socket from an HTTP server to be used as
5887 an RTSP over HTTP tunnel.
5889 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5891 * gst/rtsp-server/rtsp-client.c:
5892 * gst/rtsp-server/rtsp-media.c:
5893 * gst/rtsp-server/rtsp-media.h:
5894 rtsp: Handle the blocksize parameter
5895 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5897 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5899 * tests/check/Makefile.am:
5900 * tests/check/gst/rtspserver.c:
5901 Have unit test get header from source dir, not installed dir
5902 This makes compilation of unit tests work in a build directory other
5903 than the source directory.
5904 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5906 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5908 * gst/rtsp-server/rtsp-media.c:
5909 rtsp-media: update for gst_element_make_from_uri() changes
5911 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5914 * tests/Makefile.am:
5915 * tests/check/Makefile.am:
5916 * tests/check/gst/rtspserver.c:
5918 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5920 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5922 * gst/rtsp-server/rtsp-media.c:
5923 rtsp-media: don't collect media stats when going to NULL
5924 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5926 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5928 * gst/rtsp-server/rtsp-client.c:
5929 client: don't leak transports
5931 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5933 * gst/rtsp-server/rtsp-client.c:
5934 rtsp-client: free transport on no_stream in SETUP handler
5936 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5938 * gst/rtsp-server/rtsp-client.c:
5939 rtsp-client: changed session media iteration
5940 In client_unlink_session: now don't iterate in session->medias
5941 list where items are removed by gst_rtsp_session_release_media.
5942 Instead, repeatedly remove the first item.
5944 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5946 * gst/rtsp-server/rtsp-client.c:
5947 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5948 GstRTSPSessionMedia is not a GObject type. When the
5949 GstRTSPSession is freed, it will free the media.
5951 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5953 * gst/rtsp-server/rtsp-media-factory.c:
5954 factory: plug pad leak in collect_streams
5955 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5956 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5957 will take one reference, and the other reference will otherwise
5960 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5963 configure: suppress some warnings when debug is disabled
5964 Warnings about unused variables should be suppressed if core has the
5965 debug system disabled.
5966 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5968 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5970 * docs/libs/Makefile.am:
5971 docs: fix build in uninstalled setup
5972 Include gst-plugins-base libs properly.
5974 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5976 * docs/libs/gst-rtsp-server.types:
5977 docs: include headers defining rtsp-server object types
5978 Fixes compiler warnings during docs build.
5979 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5981 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5984 configure: Add warning flags for compiler when configuring
5985 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5987 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5990 Automatic update of common submodule
5991 From 03a0e57 to 98e386f
5993 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5996 Automatic update of common submodule
5997 From 1fab359 to 03a0e57
5999 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6001 * gst/rtsp-server/rtsp-client.c:
6002 client: fix GSocketAddress leak in gst_rtsp_client_accept
6003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6005 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6008 Automatic update of common submodule
6009 From f1b5a96 to 1fab359
6011 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6014 Automatic update of common submodule
6015 From 92b7266 to f1b5a96
6017 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6020 Automatic update of common submodule
6021 From ec1c4a8 to 92b7266
6023 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6026 Automatic update of common submodule
6027 From 3429ba6 to ec1c4a8
6029 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6031 * gst/rtsp-server/rtsp-auth.c:
6032 * gst/rtsp-server/rtsp-client.c:
6033 * gst/rtsp-server/rtsp-media-factory-uri.c:
6034 * gst/rtsp-server/rtsp-server.c:
6035 rtsp: fix compiler warnings
6036 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6038 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6041 Automatic update of common submodule
6042 From dc70203 to 3429ba6
6044 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6046 * gst/rtsp-server/rtsp-client.c:
6047 * gst/rtsp-server/rtsp-media-factory.c:
6048 * gst/rtsp-server/rtsp-media-factory.h:
6049 * gst/rtsp-server/rtsp-media.c:
6050 * gst/rtsp-server/rtsp-media.h:
6051 * gst/rtsp-server/rtsp-server.c:
6052 * gst/rtsp-server/rtsp-server.h:
6053 * gst/rtsp-server/rtsp-session-pool.c:
6054 * gst/rtsp-server/rtsp-session-pool.h:
6055 rtsp-server: port to new thread API
6057 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6060 Automatic update of common submodule
6061 From 6db25be to dc70203
6063 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6065 * gst/rtsp-server/rtsp-auth.c:
6066 * gst/rtsp-server/rtsp-auth.h:
6067 * gst/rtsp-server/rtsp-client.c:
6068 rtsp-server: Fix compilation and compiler warnings
6070 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6074 * gst/rtsp-server/Makefile.am:
6075 configure: Modernize autotools setup a bit
6076 Also we now only create tar.bz2 and tar.xz tarballs.
6078 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6081 Automatic update of common submodule
6082 From 464fe15 to 6db25be
6084 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6087 Automatic update of common submodule
6088 From 7fda524 to 464fe15
6090 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6093 * docs/libs/Makefile.am:
6094 * docs/version.entities.in:
6096 * gst/rtsp-server/Makefile.am:
6097 * pkgconfig/Makefile.am:
6098 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6099 * pkgconfig/gstreamer-rtsp-server.pc.in:
6100 * tests/Makefile.am:
6101 rtsp-server: Update versioning
6103 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6105 Merge remote-tracking branch 'origin/0.10'
6107 gst/rtsp-server/rtsp-session-pool.c
6109 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6111 * gst/rtsp-server/rtsp-session-pool.c:
6112 rtsp-server: Don't use deprecated GLib API
6114 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6116 Replace master with 0.11
6118 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6120 Merge branch 'master' into 0.11
6122 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6124 Merge branch 'master' into 0.11
6126 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6129 A couple minor typo fixes
6131 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6133 * gst/rtsp-server/rtsp-media.c:
6134 media: fix state of the appqueue
6136 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6138 * gst/rtsp-server/rtsp-media-factory-uri.c:
6139 factory: use videoconvert
6141 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6143 * gst/rtsp-server/rtsp-media-factory-uri.c:
6144 factory: change to new style caps
6146 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6148 * gst/rtsp-server/rtsp-client.c:
6149 * gst/rtsp-server/rtsp-client.h:
6150 * gst/rtsp-server/rtsp-media-factory-uri.c:
6151 * gst/rtsp-server/rtsp-media.c:
6152 * gst/rtsp-server/rtsp-server.c:
6153 * gst/rtsp-server/rtsp-server.h:
6154 * gst/rtsp-server/rtsp-session-pool.c:
6155 rtsp-server: port to GIO
6158 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6161 configure: fix build
6163 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6166 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6167 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6169 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6172 * examples/Makefile.am:
6173 First rule of gst-rtsp-server club: don't talk about gst-phonon
6175 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6178 * pkgconfig/Makefile.am:
6179 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6180 * pkgconfig/gst-rtsp-server.pc.in:
6181 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6182 * pkgconfig/gstreamer-rtsp-server.pc.in:
6183 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6184 For consistency with all other modules.
6186 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6188 * gst/rtsp-server/rtsp-client.c:
6189 rtsp-client: update for new map API
6191 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6194 * bindings/Makefile.am:
6195 * bindings/python/Makefile.am:
6196 * bindings/python/arg-types.py:
6197 * bindings/python/codegen/Makefile.am:
6198 * bindings/python/codegen/__init__.py:
6199 * bindings/python/codegen/argtypes.py:
6200 * bindings/python/codegen/code-coverage.py:
6201 * bindings/python/codegen/codegen.py:
6202 * bindings/python/codegen/definitions.py:
6203 * bindings/python/codegen/defsparser.py:
6204 * bindings/python/codegen/docextract.py:
6205 * bindings/python/codegen/docgen.py:
6206 * bindings/python/codegen/fileprefix.override:
6207 * bindings/python/codegen/fileprefixmodule.c:
6208 * bindings/python/codegen/h2def.py:
6209 * bindings/python/codegen/mergedefs.py:
6210 * bindings/python/codegen/mkskel.py:
6211 * bindings/python/codegen/override.py:
6212 * bindings/python/codegen/reversewrapper.py:
6213 * bindings/python/codegen/scmexpr.py:
6214 * bindings/python/rtspserver-types.defs:
6215 * bindings/python/rtspserver.defs:
6216 * bindings/python/rtspserver.override:
6217 * bindings/python/rtspservermodule.c:
6218 * bindings/python/test.py:
6220 python: remove pygst-based python bindings
6221 pygi is the future, apparently.
6223 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6226 Automatic update of common submodule
6227 From c463bc0 to 7fda524
6229 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6232 Automatic update of common submodule
6233 From 2a59016 to c463bc0
6235 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6238 Automatic update of common submodule
6239 From 0807187 to 2a59016
6241 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6244 Automatic update of common submodule
6245 From 11f0cd5 to 0807187
6247 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6249 * examples/test-auth.c:
6250 example: update for new caps
6252 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6254 * examples/test-video.c:
6255 * gst/rtsp-server/rtsp-client.c:
6256 * gst/rtsp-server/rtsp-media-factory-uri.c:
6257 * gst/rtsp-server/rtsp-media.c:
6258 * gst/rtsp-server/rtsp-media.h:
6259 * gst/rtsp-server/rtsp-session.c:
6260 * gst/rtsp-server/rtsp-session.h:
6261 rtsp-server: port some more to 0.11
6263 Remove bufferlist stuff
6265 Add queue before appsink now that preroll-queue-len is gone.
6266 Update for request pad changes.
6268 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6270 Merge branch 'master' into 0.11
6272 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6274 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6275 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6276 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6278 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6280 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6281 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6282 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6284 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6286 Merge branch 'master' into 0.11
6288 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-media.c:
6291 * gst/rtsp-server/rtsp-media.h:
6292 media: add a seekable boolean
6293 Maintain the seekable state with a new variable instead of reusing the
6296 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6298 * gst/rtsp-server/rtsp-media.c:
6299 Disallow seek in live media
6301 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6303 Merge branch 'master' into 0.11
6305 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6307 * gst/rtsp-server/rtsp-server.c:
6308 #ifdef statements for windows socket creation were missing
6310 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6313 Automatic update of common submodule
6314 From a39eb83 to 11f0cd5
6316 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6319 Automatic update of common submodule
6320 From 605cd9a to a39eb83
6322 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6324 Merge branch 'master' into 0.11
6326 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6328 * gst/rtsp-server/rtsp-client.c:
6329 client: use method to access property
6331 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6333 * gst/rtsp-server/rtsp-media-factory.c:
6334 * gst/rtsp-server/rtsp-media-factory.h:
6335 media-factory: add protocols property
6336 Add a property to configure the allowed protocols in the media created from the
6339 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6341 * gst/rtsp-server/rtsp-media-factory.c:
6342 * gst/rtsp-server/rtsp-media-factory.h:
6343 media-factory: add media-configure signal
6344 Add signal to allow the application to configure the media after it was created
6347 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6349 * gst/rtsp-server/rtsp-client.c:
6350 client: use method to access property
6352 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6354 * gst/rtsp-server/rtsp-media-factory.c:
6355 * gst/rtsp-server/rtsp-media-factory.h:
6356 media-factory: add protocols property
6357 Add a property to configure the allowed protocols in the media created from the
6360 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6362 * gst/rtsp-server/rtsp-media-factory.c:
6363 * gst/rtsp-server/rtsp-media-factory.h:
6364 media-factory: add media-configure signal
6365 Add signal to allow the application to configure the media after it was created
6368 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6370 Merge branch 'master' into 0.11
6372 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6374 * gst/rtsp-server/rtsp-client.c:
6375 client: use media multicast group
6377 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6379 * gst/rtsp-server/rtsp-media-factory.h:
6380 * gst/rtsp-server/rtsp-server.h:
6381 * gst/rtsp-server/rtsp-session-pool.h:
6382 * gst/rtsp-server/rtsp-session.h:
6385 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6387 * gst/rtsp-server/rtsp-client.c:
6388 * gst/rtsp-server/rtsp-sdp.h:
6389 sdp: copy and free the server ip address
6390 Copy and free the server ip address to make memory management easier later.
6392 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6394 * gst/rtsp-server/rtsp-media-factory.c:
6395 media-factory: configure multicast in media
6397 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6399 * gst/rtsp-server/rtsp-media.c:
6400 * gst/rtsp-server/rtsp-media.h:
6401 media: add property for multicast group
6402 Add a property to configure the multicast group in the media.
6403 Based on patches from Marc Leeman and Robert Krakora.
6405 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6407 * gst/rtsp-server/rtsp-media-factory.c:
6408 * gst/rtsp-server/rtsp-media-factory.h:
6409 media-factory: add property for multicast group
6410 Add a property to configure the multicast group in the media factory.
6411 Based on patches from Marc Leeman and Robert Krakora.
6413 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6415 * gst/rtsp-server/rtsp-client.c:
6416 client: do configuration of transport in one place
6417 Move the configuration of the transport destination address to where we also
6418 configure the other bits.
6420 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6422 * gst/rtsp-server/rtsp-client.c:
6423 client: use media multicast group
6425 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6427 * gst/rtsp-server/rtsp-media-factory.h:
6428 * gst/rtsp-server/rtsp-server.h:
6429 * gst/rtsp-server/rtsp-session-pool.h:
6430 * gst/rtsp-server/rtsp-session.h:
6433 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6435 * gst/rtsp-server/rtsp-client.c:
6436 * gst/rtsp-server/rtsp-sdp.h:
6437 sdp: copy and free the server ip address
6438 Copy and free the server ip address to make memory management easier later.
6440 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6442 * gst/rtsp-server/rtsp-media-factory.c:
6443 media-factory: configure multicast in media
6445 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6447 * gst/rtsp-server/rtsp-media.c:
6448 * gst/rtsp-server/rtsp-media.h:
6449 media: add property for multicast group
6450 Add a property to configure the multicast group in the media.
6451 Based on patches from Marc Leeman and Robert Krakora.
6453 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6455 * gst/rtsp-server/rtsp-media-factory.c:
6456 * gst/rtsp-server/rtsp-media-factory.h:
6457 media-factory: add property for multicast group
6458 Add a property to configure the multicast group in the media factory.
6459 Based on patches from Marc Leeman and Robert Krakora.
6461 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6463 * gst/rtsp-server/rtsp-client.c:
6464 client: do configuration of transport in one place
6465 Move the configuration of the transport destination address to where we also
6466 configure the other bits.
6468 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6470 Merge branch 'master' into 0.11
6472 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6474 * gst/rtsp-server/rtsp-client.c:
6475 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6476 The problem occurs when the client abruptly closes the connection without
6477 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6478 server is where the pipeline gets torn down. Since this handler is not called,
6479 the pipeline remains and is up and running. Subsequent clients get their own
6480 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6481 remain up and running. This is a resource leak.
6483 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6485 Merge branch 'master' into 0.11
6487 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6489 * gst/rtsp-server/rtsp-media-factory.c:
6490 * gst/rtsp-server/rtsp-media-factory.h:
6491 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6492 For example, it can be used to retrieve source elements like appsrc, in a more
6493 convenient way than subclassing get_element.
6495 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6497 Merge branch 'master' into 0.11
6499 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6501 * gst/rtsp-server/rtsp-server.c:
6502 rtsp-server: hold on to reference while using object
6504 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6506 * gst/rtsp-server/rtsp-media.c:
6509 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6512 configure: use unstable api
6514 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6516 * gst/rtsp-server/rtsp-client.c:
6517 client: fix reference counting
6519 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6521 * gst/rtsp-server/rtsp-client.c:
6522 * gst/rtsp-server/rtsp-media.c:
6523 fix compiler warnings about unused variables
6525 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6527 * examples/test-launch.c:
6528 * examples/test-readme.c:
6529 * examples/test-uri.c:
6530 * examples/test-video.c:
6531 examples: tell rtsp uri when ready
6533 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6536 Automatic update of common submodule
6537 From 69b981f to 605cd9a
6539 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6541 * gst/rtsp-server/rtsp-client.c:
6542 client: update for buffer API change
6544 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6546 * gst/rtsp-server/Makefile.am:
6547 Makefile.am: 0.10 => @GST_MAJORMINOR@
6549 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6551 * gst/rtsp-server/rtsp-media-factory-uri.c:
6552 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6554 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6556 * gst/rtsp-server/.gitignore:
6557 .gitignore: 0.10 => 0.11
6559 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6561 * gst/rtsp-server/Makefile.am:
6562 Makefile.am: 0.10 => @GST_MAJORMINOR@
6564 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6566 Merge branch 'master' into 0.11
6568 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6571 Automatic update of common submodule
6572 From 9e5bbd5 to 69b981f
6574 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6577 Automatic update of common submodule
6578 From fd35073 to 9e5bbd5
6580 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6583 Automatic update of common submodule
6584 From 46dfcea to fd35073
6586 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6588 * gst/rtsp-server/rtsp-media-factory-uri.c:
6589 * gst/rtsp-server/rtsp-media.c:
6590 media: port to new caps API
6592 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6594 Merge branch 'master' into 0.11
6596 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6598 * bindings/vala/gst-rtsp-server-0.10.vapi:
6599 Updated Vala bindings.
6600 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6602 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6604 * gst/rtsp-server/rtsp-server.c:
6605 * gst/rtsp-server/rtsp-server.h:
6606 Add a signal for newly connected clients.
6607 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6609 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6611 * bindings/python/rtspserver.override:
6612 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6614 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * gst/rtsp-server/Makefile.am:
6617 * gst/rtsp-server/rtsp-client.c:
6618 * gst/rtsp-server/rtsp-funnel.c:
6619 * gst/rtsp-server/rtsp-funnel.h:
6620 * gst/rtsp-server/rtsp-media.c:
6621 rtsp-server: port to 0.11
6623 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6628 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6630 Merge branch 'master' into 0.11
6635 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6638 Automatic update of common submodule
6639 From c3cafe1 to 46dfcea
6641 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6643 * bindings/python/Makefile.am:
6644 * bindings/python/rtspserver.defs:
6645 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6647 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6649 * bindings/python/arg-types.py:
6650 python bindings: add GstRTSPUrlParam
6651 Needed to implement MediaFactory virtual proxies
6653 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6655 * bindings/python/arg-types.py:
6656 python bindings: fix returning GstRTSPUrl types
6658 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6660 * bindings/python/arg-types.py:
6661 python bindings: add arg type for GstRTSPUrl
6663 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6665 * bindings/python/rtspserver.defs:
6666 python bindings: fix the definition of MediaFactory.collect_stream
6668 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6671 Automatic update of common submodule
6672 From 1ccbe09 to c3cafe1
6674 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6677 Automatic update of common submodule
6678 From 193b717 to 1ccbe09
6680 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6683 Automatic update of common submodule
6684 From b77e2bf to 193b717
6686 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6689 build: Include lcov.mak to allow test coverage report generation
6691 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6694 Automatic update of common submodule
6695 From d8814b6 to b77e2bf
6697 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6700 Automatic update of common submodule
6701 From 6aaa286 to d8814b6
6703 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6706 Automatic update of common submodule
6707 From 6aec6b9 to 6aaa286
6709 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6712 autogen: wingo signed comment
6714 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6716 * gst/rtsp-server/rtsp-session-pool.c:
6717 session: use full charset for RTSP session ID
6718 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6719 session ID more difficult.
6720 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6722 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6724 * gst/rtsp-server/Makefile.am:
6725 rtsp-server: Don't install the funnel header
6727 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6730 Automatic update of common submodule
6731 From 1de7f6a to 6aec6b9
6733 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6736 configure: require core/base 0.10.31
6737 Needed at least for gst_plugin_feature_rank_compare_func().
6739 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6742 Automatic update of common submodule
6743 From f94d739 to 1de7f6a
6745 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6747 * gst/rtsp-server/rtsp-media.c:
6748 media: remove more unused code
6750 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6752 * gst/rtsp-server/rtsp-media.c:
6753 * gst/rtsp-server/rtsp-media.h:
6754 media: remove duplicate filtering
6755 Remove the duplicate filtering code now that we have a released -good version.
6756 Give a warning instead.
6758 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6760 * gst/rtsp-server/rtsp-media-factory.c:
6761 * gst/rtsp-server/rtsp-media.c:
6762 media: fix default buffer size
6764 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6766 * gst/rtsp-server/rtsp-media-factory.c:
6767 * gst/rtsp-server/rtsp-media-factory.h:
6768 media-factory: add property to configure the buffer-size
6769 Add a property to configure the kernel UDP buffer size.
6771 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6773 * gst/rtsp-server/rtsp-media.c:
6774 * gst/rtsp-server/rtsp-media.h:
6775 media: add property to configure kernel buffer sizes
6776 Add a property to configure the kernel UDP buffer size.
6778 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6781 configure: set PYGOBJECT_REQ before using it
6782 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6784 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6787 docs: recursive into sub-directories on 'make upload'
6789 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6791 * docs/libs/gst-rtsp-server-docs.sgml:
6792 * docs/version.entities.in:
6793 docs: mention full version these docs are for, not just major-minor
6795 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6800 === release 0.10.8 ===
6802 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6807 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6809 * gst/rtsp-server/rtsp-server.c:
6810 rtsp-server: clarify docs a little
6812 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6814 * gst/rtsp-server/rtsp-media.c:
6815 media: init debug category before starting thread
6817 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6819 * gst/rtsp-server/rtsp-auth.c:
6820 auth: add realm to make it more spec compliant
6822 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6824 * gst/rtsp-server/rtsp-server.c:
6825 * gst/rtsp-server/rtsp-server.h:
6828 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6830 * examples/test-video.c:
6831 example: improve example docs a little
6833 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6835 * gst/rtsp-server/rtsp-server.c:
6836 server: ensure the watch has a ref to the server
6838 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6840 * gst/rtsp-server/rtsp-server.c:
6841 server: simpify channel function
6843 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6845 * gst/rtsp-server/rtsp-server.c:
6846 * gst/rtsp-server/rtsp-server.h:
6847 server: simplify management of channel and source
6848 We don't need to keep around the channel and source objects. Let the mainloop
6849 and the source manage the source and channel respectively.
6851 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6857 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6860 * tests/Makefile.am:
6861 * tests/test-cleanup.c:
6862 tests: add tests directory and cleanup test
6864 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6866 * gst/rtsp-server/rtsp-media-factory-uri.c:
6867 * gst/rtsp-server/rtsp-media-factory.c:
6868 * gst/rtsp-server/rtsp-media-mapping.c:
6869 * gst/rtsp-server/rtsp-media.c:
6870 * gst/rtsp-server/rtsp-session-pool.c:
6871 * gst/rtsp-server/rtsp-session.c:
6872 server: improve debugging in various objects
6874 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6876 * gst/rtsp-server/rtsp-server.c:
6877 server: chain up to the parent finalize
6879 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6881 * bindings/python/rtspserver-types.defs:
6882 * bindings/python/rtspserver.defs:
6883 * bindings/python/rtspserver.override:
6884 * bindings/python/test.py:
6885 gst-rtsp-server: update python bindings
6887 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6889 * gst/rtsp-server/rtsp-client.c:
6890 client: use the response from the clientstate
6891 Create the response object only once and store in the client state.
6892 Make all methods use the state response,
6894 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6896 * gst/rtsp-server/rtsp-server.c:
6897 server: use signal to keep track of clients
6898 Keep track of all the clients that the server creates and remove them when they
6899 fire the 'closed' signal.
6901 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6903 * gst/rtsp-server/rtsp-client.c:
6904 * gst/rtsp-server/rtsp-client.h:
6905 client: emit signal when closing
6907 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6909 * examples/.gitignore:
6910 * examples/Makefile.am:
6911 * examples/test-auth.c:
6912 * examples/test-video.c:
6913 * gst/rtsp-server/rtsp-auth.c:
6914 * gst/rtsp-server/rtsp-auth.h:
6915 * gst/rtsp-server/rtsp-client.c:
6916 * gst/rtsp-server/rtsp-media-factory.c:
6917 * gst/rtsp-server/rtsp-media.c:
6918 * gst/rtsp-server/rtsp-media.h:
6919 * gst/rtsp-server/rtsp-session-pool.h:
6920 * gst/rtsp-server/rtsp-session.h:
6921 media: enable per factory authorisations
6922 Allow for adding a GstRTSPAuth on the factory and media level and check
6923 permissions when accessing the factory.
6924 Add hints to the auth methods for future more fine grained authorisation.
6925 Add example application for per factory authentication.
6927 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6929 * gst/rtsp-server/rtsp-auth.c:
6930 * gst/rtsp-server/rtsp-auth.h:
6931 * gst/rtsp-server/rtsp-client.c:
6932 * gst/rtsp-server/rtsp-client.h:
6933 * gst/rtsp-server/rtsp-params.c:
6934 * gst/rtsp-server/rtsp-params.h:
6935 rtsp-server: Pass ClientState structure arround
6936 Pass the collected information for the ongoing request in a GstRTSPClientState
6937 structure that we can then pass around to simplify the method arguments. This
6938 will also be handy when we implement logging functionality.
6940 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6942 * gst/rtsp-server/rtsp-media-factory.c:
6943 * gst/rtsp-server/rtsp-media-factory.h:
6944 media-factory: add methods to configure authorisation
6946 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6948 * gst/rtsp-server/rtsp-client.c:
6949 client: unref auth in finalize
6951 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6953 * gst/rtsp-server/rtsp-server.c:
6954 server: unref auth in finalize
6956 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6958 * docs/libs/gst-rtsp-server-docs.sgml:
6959 * docs/libs/gst-rtsp-server-sections.txt:
6960 * docs/libs/gst-rtsp-server.types:
6963 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6965 * gst/rtsp-server/rtsp-server.c:
6966 * gst/rtsp-server/rtsp-server.h:
6967 server: separate create and accept
6968 Create separate create and accept methods so that subclasses can create custom
6970 Configure the server in the client object and prepare for keeping track of
6973 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6975 * gst/rtsp-server/rtsp-client.c:
6976 * gst/rtsp-server/rtsp-client.h:
6977 client: add support for setting the server.
6978 Add support for keeping a ref to the server that started this client
6981 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6983 * gst/rtsp-server/rtsp-auth.c:
6984 auth: fix memleak and add some docs
6985 Fix a memleak of the basic auth token.
6986 Add docs for the helper function
6988 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6990 * gst/rtsp-server/rtsp-auth.c:
6991 * gst/rtsp-server/rtsp-auth.h:
6992 * gst/rtsp-server/rtsp-client.c:
6993 client: delegate setup of auth to the manager
6994 Delegate the configuration of the authentication tokens to the manager object
6997 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6999 * examples/test-video.c:
7000 * gst/rtsp-server/Makefile.am:
7001 * gst/rtsp-server/rtsp-auth.c:
7002 * gst/rtsp-server/rtsp-auth.h:
7003 * gst/rtsp-server/rtsp-client.c:
7004 * gst/rtsp-server/rtsp-client.h:
7005 * gst/rtsp-server/rtsp-server.c:
7006 * gst/rtsp-server/rtsp-server.h:
7007 auth: add authentication object
7008 Add an object that can check the authorization of requests.
7009 Implement basic authentication.
7010 Add example authentication to test-video
7012 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7014 * gst/rtsp-server/rtsp-server.c:
7015 * gst/rtsp-server/rtsp-server.h:
7016 server: move includes back
7017 the includes are needed for sockaddr_in.
7019 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7021 * gst/rtsp-server/rtsp-client.c:
7022 * gst/rtsp-server/rtsp-client.h:
7023 * gst/rtsp-server/rtsp-server.c:
7024 * gst/rtsp-server/rtsp-server.h:
7025 rtsp: move network includes where they are needed
7027 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7029 * gst/rtsp-server/rtsp-media.h:
7030 rtsp-media.h: Minor corrections in comments.
7033 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7036 Automatic update of common submodule
7037 From e572c87 to f94d739
7039 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7043 * docs/libs/.gitignore:
7044 * examples/.gitignore:
7045 * gst/rtsp-server/.gitignore:
7048 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7050 * docs/libs/Makefile.am:
7051 docs: We don't build ps/pdf for API reference docs
7053 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7056 Automatic update of common submodule
7057 From ccbaa85 to e572c87
7059 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7062 Automatic update of common submodule
7063 From 46445ad to ccbaa85
7065 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7067 * gst/rtsp-server/Makefile.am:
7068 * gst/rtsp-server/fs-funnel.c:
7069 * gst/rtsp-server/fs-funnel.h:
7070 * gst/rtsp-server/rtsp-funnel.c:
7071 * gst/rtsp-server/rtsp-funnel.h:
7072 * gst/rtsp-server/rtsp-media.c:
7073 funnel: rename fsfunnel to rtspfunnel
7074 Rename the funnel to avoid conflicts with the farsight one.
7076 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7078 * gst/rtsp-server/Makefile.am:
7079 * gst/rtsp-server/fs-funnel.c:
7080 * gst/rtsp-server/fs-funnel.h:
7081 * gst/rtsp-server/rtsp-media.c:
7082 rtsp-media: add and use fsfunnel
7083 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7084 select-all property that we need.
7086 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7088 * gst/rtsp-server/Makefile.am:
7089 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7090 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7091 for the g-ir-compiler, rather than just assuming the env var has
7094 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7101 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7103 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7106 * gst/rtsp-server/Makefile.am:
7107 gobject-introspection: fix g-i build for uninstalled setup
7108 Requires gst-plugins-base git (> 0.10.31.2).
7110 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7112 * examples/test-uri.c:
7113 examples: add some more options and comments
7115 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7117 * gst/rtsp-server/rtsp-media-factory-uri.c:
7118 factory-uri: use right property type
7120 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7122 * gst/rtsp-server/rtsp-media-factory-uri.c:
7123 factory-uri: attempt to configure buffer-lists
7124 Attempt to configure buffer lists in the payloader for improved performance.
7126 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-media.c:
7129 media: attempt to configure bigger UDP buffers
7130 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7131 send buffers with high bitrate streams.
7133 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7135 * gst/rtsp-server/rtsp-client.c:
7136 client: use the socket length from getsockname
7137 Use the length returned by getsockname to perform the getnameinfo call because
7138 the size can depend on the socket type and platform.
7141 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7143 * docs/libs/gst-rtsp-server-docs.sgml:
7144 * docs/libs/gst-rtsp-server-sections.txt:
7145 docs: add uri factory to the docs
7147 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7149 * gst/rtsp-server/rtsp-client.c:
7150 * gst/rtsp-server/rtsp-media.h:
7153 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7155 * gst/rtsp-server/rtsp-client.c:
7156 * gst/rtsp-server/rtsp-media.c:
7157 * gst/rtsp-server/rtsp-media.h:
7158 * gst/rtsp-server/rtsp-session.c:
7159 * gst/rtsp-server/rtsp-session.h:
7160 rtsp-server: add support for buffer lists
7161 Add support for sending bufferlists received from appsink.
7164 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7166 * gst/rtsp-server/rtsp-client.c:
7167 * gst/rtsp-server/rtsp-media.c:
7168 * gst/rtsp-server/rtsp-media.h:
7169 * gst/rtsp-server/rtsp-sdp.c:
7170 media: make method to retrieve the play range
7171 Make a method to retrieve the playback range so that we can conditionally create
7172 a different range for the SDP and the PLAY requests.
7174 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-media.c:
7177 * gst/rtsp-server/rtsp-media.h:
7178 media: add signal to notify of state changes
7180 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7182 * gst/rtsp-server/rtsp-client.h:
7183 client: cleanup headers
7185 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * gst/rtsp-server/rtsp-client.c:
7190 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7192 * gst/rtsp-server/rtsp-media-factory-uri.c:
7193 * gst/rtsp-server/rtsp-media-factory-uri.h:
7194 factory-uri: add support for gstpay
7195 Add an option to prefer gstpay over decoder + raw payloader.
7197 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7199 * gst/rtsp-server/rtsp-media-factory-uri.c:
7200 * gst/rtsp-server/rtsp-media-factory-uri.h:
7201 factory-uri: rework the autoplugger.
7202 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7205 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7207 * gst/rtsp-server/rtsp-media-factory-uri.c:
7208 factory-uri: use better factory filter
7209 Make better payloader filter based on autoplug rank and RTP use case.
7211 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7214 Automatic update of common submodule
7215 From 169462a to 46445ad
7217 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7219 * gst/rtsp-server/rtsp-server.c:
7220 server: set SO_REUSEADDR before bind
7221 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7223 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7225 * gst/rtsp-server/rtsp-media.c:
7226 * gst/rtsp-server/rtsp-media.h:
7227 media: emit prepared signal when prepared
7228 Make a 'prepared' signal and emit it when we successfully prepared the element.
7229 This signal can be used to configure the media object after it has been prepared
7232 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7235 Automatic update of common submodule
7236 From 011bcc8 to 169462a
7238 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7240 python an optional dependency
7241 * configure.ac: Move up valgrind and g-i checks. Make the python
7242 dependency optional, as it was before.
7244 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7246 Merge branch 'master' into 0.11
7251 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-media.c:
7254 media: update range when active clients changed
7255 When we changed the number of active clients, update the current range
7256 information because we want the second client connecting to a shared resource
7257 continue from where the stream currently.
7259 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7261 * gst/rtsp-server/rtsp-media-factory-uri.c:
7262 * gst/rtsp-server/rtsp-media-factory-uri.h:
7263 factory-uri: add colorspace and fix pt
7264 Rework the way we pass data to the autoplugger.
7265 When we have raw caps, plug a converter element to make pluggin to raw
7266 payloaders more successful.
7267 Make sure all dynamically plugged payloaders have a unique payload types.
7269 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7271 * examples/Makefile.am:
7272 * examples/test-uri.c:
7273 example: add example of the uri factory
7275 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7277 * gst/rtsp-server/Makefile.am:
7278 * gst/rtsp-server/rtsp-media-factory-uri.c:
7279 * gst/rtsp-server/rtsp-media-factory-uri.h:
7280 * gst/rtsp-server/rtsp-server.h:
7281 factory-uri: add a factory to stream any URI
7282 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7285 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7287 * gst/rtsp-server/rtsp-media.c:
7288 * gst/rtsp-server/rtsp-media.h:
7289 media: ignore spurious ASYNC_DONE messages
7290 When we are dynamically adding pads, the addition of the udpsrc elements will
7291 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7292 the real ASYNC_DONE when everything is prerolled.
7294 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7296 * gst/rtsp-server/rtsp-media-factory.c:
7297 * gst/rtsp-server/rtsp-media-factory.h:
7298 media-factory: make lock macro
7300 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7302 * gst/rtsp-server/rtsp-client.c:
7303 rtsp-server: Remove unused variable and dead assignment
7305 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7307 * examples/test-launch.c:
7308 * examples/test-mp4.c:
7309 * examples/test-ogg.c:
7310 * examples/test-readme.c:
7311 * examples/test-sdp.c:
7312 * examples/test-video.c:
7313 examples: Run gst-indent
7315 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7317 * gst/rtsp-server/rtsp-client.c:
7318 * gst/rtsp-server/rtsp-media-factory.c:
7319 * gst/rtsp-server/rtsp-media-mapping.c:
7320 * gst/rtsp-server/rtsp-media.c:
7321 * gst/rtsp-server/rtsp-params.c:
7322 * gst/rtsp-server/rtsp-sdp.c:
7323 * gst/rtsp-server/rtsp-server.c:
7324 * gst/rtsp-server/rtsp-session-pool.c:
7325 * gst/rtsp-server/rtsp-session.c:
7326 rtsp-server: Run gst-indent
7327 Since it wasn't using the upstream common previously, there was no
7328 indentation check before commiting.
7330 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7332 * gst/rtsp-server/rtsp-media-mapping.h:
7333 * gst/rtsp-server/rtsp-media.c:
7334 * gst/rtsp-server/rtsp-media.h:
7335 * gst/rtsp-server/rtsp-sdp.c:
7336 * gst/rtsp-server/rtsp-session-pool.h:
7337 * gst/rtsp-server/rtsp-session.c:
7338 * gst/rtsp-server/rtsp-session.h:
7339 rtsp-server: Some more doc fixups
7341 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7344 Makefile: Add cruft-cleaning support
7346 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7351 * docs/libs/Makefile.am:
7352 * docs/libs/gst-rtsp-server-docs.sgml:
7353 * docs/libs/gst-rtsp-server-sections.txt:
7354 * docs/libs/gst-rtsp-server.types:
7355 * docs/version.entities.in:
7356 docs: Add gtk-doc build system
7358 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7360 * gst/rtsp-server/Makefile.am:
7361 Makefile.am: Use standard GIR make behaviour
7363 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7367 autogen/configure: Bring more in sync to standard gst module behaviour
7369 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7371 * gst/rtsp-server/rtsp-media.c:
7372 media: warn and fail when gstrtpbin is not found
7374 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7377 configure: open 0.11 branch
7379 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7383 Add common submodule
7385 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7388 * common/Makefile.am:
7389 * common/c-to-xml.py:
7391 * common/coverage/coverage-report-entry.pl:
7392 * common/coverage/coverage-report.pl:
7393 * common/coverage/coverage-report.xsl:
7394 * common/coverage/lcov.mak:
7395 * common/gettext.patch:
7396 * common/glib-gen.mak:
7397 * common/gst-autogen.sh:
7398 * common/gst-xmlinspect.py:
7400 * common/gstdoc-scangobj:
7401 * common/gtk-doc-plugins.mak:
7402 * common/gtk-doc.mak:
7403 * common/m4/.gitignore:
7404 * common/m4/Makefile.am:
7406 * common/m4/as-ac-expand.m4:
7407 * common/m4/as-auto-alt.m4:
7408 * common/m4/as-compiler-flag.m4:
7409 * common/m4/as-compiler.m4:
7410 * common/m4/as-docbook.m4:
7411 * common/m4/as-libtool-tags.m4:
7412 * common/m4/as-libtool.m4:
7413 * common/m4/as-python.m4:
7414 * common/m4/as-scrub-include.m4:
7415 * common/m4/as-version.m4:
7416 * common/m4/ax_create_stdint_h.m4:
7417 * common/m4/check.m4:
7418 * common/m4/glib-gettext.m4:
7419 * common/m4/gst-arch.m4:
7420 * common/m4/gst-args.m4:
7421 * common/m4/gst-check.m4:
7422 * common/m4/gst-debuginfo.m4:
7423 * common/m4/gst-default.m4:
7424 * common/m4/gst-doc.m4:
7425 * common/m4/gst-error.m4:
7426 * common/m4/gst-feature.m4:
7427 * common/m4/gst-function.m4:
7428 * common/m4/gst-gettext.m4:
7429 * common/m4/gst-glib2.m4:
7430 * common/m4/gst-libxml2.m4:
7431 * common/m4/gst-plugindir.m4:
7432 * common/m4/gst-valgrind.m4:
7433 * common/m4/gtk-doc.m4:
7434 * common/m4/introspection.m4:
7436 * common/mangle-tmpl.py:
7437 * common/plugins.xsl:
7439 * common/release.mak:
7440 * common/scangobj-merge.py:
7441 * common/upload.mak:
7442 common: Remove static version
7444 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7446 * common/m4/introspection.m4:
7447 Update introspection.m4 to match usage
7449 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7453 Remove old stuff from the README
7455 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7460 === release 0.10.7 ===
7462 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7467 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7469 * examples/test-ogg.c:
7470 test-ogg: remove parsers
7471 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7472 buffers with timestamps. Using the parsers also seems to break things.
7474 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7476 * bindings/vala/gst-rtsp-server-0.10.vapi:
7477 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7478 Updated Vala bindings
7480 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7482 * common/m4/introspection.m4:
7484 * gst/rtsp-server/Makefile.am:
7485 Added initial gobject-introspection support
7487 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7489 * gst/rtsp-server/rtsp-media-factory.c:
7490 media-factory: don't use host for shared hash key
7491 When we generate the key to share made between connections, don't include the
7492 host used to connect so that we can share media even if between clients that
7493 connected with localhost and ones with the ip address.
7495 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7497 * bindings/vala/Makefile.am:
7498 build: fix distcheck
7500 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7502 * bindings/vala/gst-rtsp-server-0.10.vapi:
7503 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7504 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7505 Update Vala bindings
7507 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7509 * bindings/vala/Makefile.am:
7511 Fix configure checks and installation location for Vala bindings
7514 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7519 === release 0.10.6 ===
7521 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7524 configure: release 0.10.6
7526 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7528 * gst/rtsp-server/rtsp-media.c:
7529 media: help the compiler a little
7531 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7533 * gst/rtsp-server/rtsp-media.c:
7534 * gst/rtsp-server/rtsp-media.h:
7535 * gst/rtsp-server/rtsp-session.c:
7536 media: cleanup media transport before freeing
7537 Cleanup the media transport data before freeing. In particular, remove the qdata
7538 from the rtpsource object.
7540 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7542 * gst/rtsp-server/rtsp-media-factory.c:
7543 * gst/rtsp-server/rtsp-media-factory.h:
7544 * gst/rtsp-server/rtsp-media.c:
7545 * gst/rtsp-server/rtsp-media.h:
7546 media-factory: add eos-shutdown property
7547 Add an eos-shutdown property that will send an EOS to the pipeline before
7548 shutting it down. This allows for nice cleanup in case of a muxer.
7551 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7553 * gst/rtsp-server/rtsp-media.c:
7554 * gst/rtsp-server/rtsp-media.h:
7555 media: use multiudpsink send-duplicates when we can
7556 If we have a new enough multiudpsink with the send-duplicates property, use this
7557 instead of doing our own filtering. Our custom filtering code should eventually
7558 be removed when we can depend on a released -good.
7560 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7562 * gst/rtsp-server/rtsp-media.c:
7563 media: don't leak destinations
7564 Refactor and cleanup the destinations array when the stream is destroyed.
7566 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7568 * gst/rtsp-server/rtsp-media.c:
7569 * gst/rtsp-server/rtsp-media.h:
7570 media: don't add udp addresses multiple times
7571 Keep track of the udp addresses we added to udpsink and never add the same udp
7572 destination twice. This avoids duplicate packets when using multicast.
7574 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7576 * gst/rtsp-server/rtsp-server.c:
7577 server: disable use of SO_LINGER
7578 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7579 server close()s the connection.
7581 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7583 * gst/rtsp-server/rtsp-server.c:
7584 server: use 5 second linger period in SO_LINGER
7585 Wait 5 seconds before clearing the send buffers and reseting the connection with
7586 the client when we do a close. This should be enough time to get the message to
7590 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7592 * gst/rtsp-server/rtsp-server.c:
7593 server: use SO_LINGER
7594 SO_LINGER on the socket will make sure that any pending data on the socket is
7595 flushed ASAP and that the socket connection is reset. This makes sure that the
7596 socket can be reused immediately.
7599 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7602 README: add blurb about shared media factories
7604 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7606 * gst/rtsp-server/rtsp-media.c:
7607 Add stdlib.h for atoi()
7609 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7611 * bindings/python/Makefile.am:
7612 * bindings/vala/Makefile.am:
7613 build: distcheck fixes
7614 Fix 'make distcheck', somewhat (it still fails because it tries to
7615 install files into /usr/share/vala/vapi/ irrespective of the
7618 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7621 configure: bump core/base requirements to released version
7622 Makes things less confusing for people.
7624 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7627 configure: fail if GStreamer core/base requirements are not met
7629 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7631 * gst/rtsp-server/rtsp-client.c:
7632 client: improve client cleanups
7633 Make sure the session does not timeout when using TCP. We need to do this
7634 because quicktime player does not send RTCP for some reason in tunneled
7636 Refactor some cleanup code.
7639 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7641 * gst/rtsp-server/rtsp-session.c:
7642 * gst/rtsp-server/rtsp-session.h:
7643 session: add support for prevent session timeouts
7644 Add an atomix counter to prevent session timeouts when we are, for example,
7647 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7649 * gst/rtsp-server/rtsp-client.c:
7650 client: fix unlink on session timeouts
7651 When our session times out, make sure we unlink all streams in this
7653 Remove the tunnelid when closing the connection.
7655 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7657 * gst/rtsp-server/rtsp-session.c:
7658 session: small cleanups
7660 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7662 * gst/rtsp-server/rtsp-client.c:
7663 client: handle lost_tunnel callbacks
7664 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7665 hashtable so that we can reuse it for when the client reopens the POST
7667 Close the connection after a TEARDOWN.
7668 Make sure or watchid is cleared when the watch is removed.
7671 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7673 * gst/rtsp-server/rtsp-client.c:
7674 * gst/rtsp-server/rtsp-media.c:
7675 * gst/rtsp-server/rtsp-sdp.c:
7676 rtsp-server: add more support for multicast
7678 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7681 * gst/rtsp-server/rtsp-media.c:
7682 * gst/rtsp-server/rtsp-media.h:
7683 media: allow configuration of allowed lower transport
7685 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7687 * gst/rtsp-server/rtsp-client.h:
7688 * gst/rtsp-server/rtsp-media.c:
7689 * gst/rtsp-server/rtsp-media.h:
7690 * gst/rtsp-server/rtsp-sdp.c:
7691 * gst/rtsp-server/rtsp-sdp.h:
7692 * gst/rtsp-server/rtsp-server.c:
7693 rtsp: keep track of server ip and ipv6
7694 Keep track of how the client connected to the server and setup the udp ports
7695 with the same protocol.
7696 Copy the server ip address in the SDP so that clients can send RTCP back to
7699 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7701 * gst/rtsp-server/rtsp-session.c:
7704 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7706 * gst/rtsp-server/rtsp-client.c:
7707 client: use right size for malloc
7709 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7711 * gst/rtsp-server/rtsp-server.c:
7712 server: comment ipv6 server listening address
7714 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7716 * gst/rtsp-server/rtsp-media.c:
7717 media: allow for ipv6 sockets
7719 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7721 * gst/rtsp-server/rtsp-server.c:
7722 * gst/rtsp-server/rtsp-server.h:
7723 server: rework server part
7724 Allow setting a bind address, make sure we can deal with ipv6.
7725 Remove the port property and change with the service property.
7727 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-media.h:
7730 media: update comments a little
7732 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7734 * gst/rtsp-server/rtsp-client.c:
7735 client: make content-base better
7736 Use the URI formatting functions to make a content-base. Also make sure that
7737 there is a trailing / at the end.
7739 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7741 * gst/rtsp-server/rtsp-client.c:
7742 client: guard against invalid paths
7744 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7746 * examples/test-video.c:
7747 test: catch server bind errors
7749 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7751 * gst/rtsp-server/rtsp-media.c:
7752 rtspmedia: emit "unprepared" if _prepare fails.
7753 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7754 media object is removed from its factory's cache.
7756 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7758 * gst/rtsp-server/rtsp-media.c:
7759 media: collect media position when seek completes
7761 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7763 * gst/rtsp-server/rtsp-client.c:
7764 client: call unlink_streams in client finalize
7767 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7769 * gst/rtsp-server/rtsp-media.c:
7770 media: limit the time to wait to something huge
7771 Avoid waiting forever but limit the timeout to 20 seconds.
7773 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7775 * gst/rtsp-server/rtsp-sdp.c:
7776 sdp: reindent and check for prepared status
7778 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7780 * gst/rtsp-server/rtsp-media.c:
7781 * gst/rtsp-server/rtsp-media.h:
7782 * gst/rtsp-server/rtsp-session.c:
7783 media: avoid doing _get_state() for state changes
7784 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7785 until the media is prerolled or in error. This avoids doing a blocking call of
7786 gst_element_get_state() that can cause lockups when there is an error.
7789 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7791 * gst/rtsp-server/rtsp-media.c:
7794 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7796 * gst/rtsp-server/rtsp-media-factory.c:
7797 media-factory: better error handling
7798 Improve the error handling a bit.
7800 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7802 * gst/rtsp-server/rtsp-client.c:
7803 client: rework transport parsing
7804 Rework the transport parsing code so that we can ignore transports we don't
7805 support instead of just picking the first one we can parse.
7806 Configure a (for now hardcoded) destination for multicast transports.
7808 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7810 * gst/rtsp-server/rtsp-media.c:
7811 media: set multicast sink parameters
7812 Disable loop and automatic multicast join on the udpsink elements.
7813 Add some more debug info.
7814 Reset some state variables in the right place.
7815 Use the right port numbers for multicast.
7817 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7819 * gst/rtsp-server/rtsp-session.c:
7820 session: handle transport setup correctly
7821 Handle UDP, MCAST and TCP transport negotiation more correctly.
7822 Store the server session SSRC in the transport.
7824 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7826 * gst/rtsp-server/rtsp-client.c:
7827 rtsp-client: implement error_full
7828 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7831 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7834 * gst/rtsp-server/rtsp-client.c:
7835 * gst/rtsp-server/rtsp-server.c:
7836 docs: update docs and comments
7838 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7840 * gst/rtsp-server/rtsp-sdp.c:
7841 sdp: make server work better when behind a proxy
7843 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7845 * gst/rtsp-server/rtsp-client.c:
7846 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7848 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7850 * gst/rtsp-server/rtsp-client.c:
7851 * gst/rtsp-server/rtsp-media-factory.c:
7852 * gst/rtsp-server/rtsp-media-mapping.c:
7853 * gst/rtsp-server/rtsp-media.c:
7854 * gst/rtsp-server/rtsp-server.c:
7855 * gst/rtsp-server/rtsp-session-pool.c:
7856 * gst/rtsp-server/rtsp-session.c:
7857 Use GStreamer's debugging subsystem
7859 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7861 * gst/rtsp-server/rtsp-media-factory.c:
7862 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7864 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7869 === release 0.10.5 ===
7871 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7876 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7879 configure: bump required versions
7881 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7883 * gst/rtsp-server/rtsp-client.c:
7884 client: call weak-unref on client->sessions from finalize
7887 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7889 * gst/rtsp-server/rtsp-media.c:
7890 media: Fixed crasher where caps got unref'ed too often
7892 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7895 * pkgconfig/.gitignore:
7896 * pkgconfig/Makefile.am:
7897 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7898 Added pkg-config file to use gst-rtsp-server uninstalled
7900 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7902 * gst/rtsp-server/rtsp-media.c:
7903 media: add some docs
7905 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7907 * gst/rtsp-server/rtsp-client.c:
7908 rtsp: Use gst_rtsp_watch_send_message().
7909 Use gst_rtsp_watch_send_message() since the old API which used
7910 gst_rtsp_watch_queue_message() has been deprecated.
7912 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7917 === release 0.10.4 ===
7919 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7924 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7926 * gst/rtsp-server/rtsp-client.c:
7927 * gst/rtsp-server/rtsp-session.c:
7928 * gst/rtsp-server/rtsp-session.h:
7929 rtsp: allocate channels in TCP mode
7930 When the client does not provide us with channels in TCP mode, allocate channels
7933 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7935 * gst/rtsp-server/rtsp-client.c:
7936 client: don't crash when tunnelid is missing
7937 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7938 don't crash but return an error response to the client.
7941 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7943 * bindings/vala/gst-rtsp-server-0.10.vapi:
7944 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7945 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7946 bindings: update vala bindings with new method
7948 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7950 * gst/rtsp-server/rtsp-session-pool.c:
7951 * gst/rtsp-server/rtsp-session-pool.h:
7952 sessionpool: add function to filter sessions
7953 Add generic function to retrieve/remove sessions.
7955 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7958 configure: bump core/base requirements to release
7960 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7962 * gst/rtsp-server/rtsp-media.c:
7963 media: fix indentation
7965 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7967 * gst/rtsp-server/rtsp-media.c:
7968 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7970 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7972 * gst/rtsp-server/rtsp-media.c:
7973 set state and remove elements of media in for loop
7975 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7977 * bindings/vala/gst-rtsp-server-0.10.vapi:
7978 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7979 Added gst_rtsp_media_remove_elements function to Vala bindings
7981 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7983 * gst/rtsp-server/rtsp-media.c:
7984 * gst/rtsp-server/rtsp-media.h:
7985 Added gst_rtsp_media_remove_elements function
7987 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7989 * gst/rtsp-server/rtsp-media.c:
7990 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7992 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7994 * bindings/vala/gst-rtsp-server-0.10.vapi:
7995 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7996 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7997 Updated Vala bindings
7999 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8001 * gst/rtsp-server/rtsp-media.c:
8002 * gst/rtsp-server/rtsp-media.h:
8003 Added vmethod unprepare to GstRTSPMedia
8004 The default implementation sets the state of the pipeline to GST_STATE_NULL
8006 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8008 * gst/rtsp-server/rtsp-media-factory.c:
8009 * gst/rtsp-server/rtsp-media-factory.h:
8010 Made collect_streams function public
8012 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8014 * gst/rtsp-server/rtsp-media-factory.c:
8015 * gst/rtsp-server/rtsp-media-factory.h:
8016 * gst/rtsp-server/rtsp-media.c:
8017 Added vmethod create_pipeline to GstRTSPMediaFactory
8018 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8020 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8022 * gst/rtsp-server/rtsp-client.c:
8023 client: use g_source_destroy()
8024 We need to use g_source_destroy() because we might have added the source to a
8025 different main context than the default one.
8027 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8029 * gst/rtsp-server/Makefile.am:
8030 * gst/rtsp-server/rtsp-client.c:
8031 * gst/rtsp-server/rtsp-params.c:
8032 * gst/rtsp-server/rtsp-params.h:
8033 rtsp: prepare for handling GET/SET_PARAMETER
8034 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8036 Fix return codes of handlers.
8038 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8040 * gst/rtsp-server/rtsp-media.c:
8041 media: don't leak session pads
8043 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8045 * gst/rtsp-server/rtsp-media.c:
8046 media: clean up the messages a bit
8048 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8050 * gst/rtsp-server/rtsp-sdp.c:
8051 sdp: warn and skip streams without media
8053 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8055 * bindings/vala/gst-rtsp-server-0.10.vapi:
8056 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8057 vala: Fixed typo in header file of RTSPMediaStream
8059 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8061 * gst/rtsp-server/rtsp-media.c:
8064 Make dumping RTCP stats configurable
8066 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8068 * gst/rtsp-server/rtsp-media.c:
8069 media: be less verbose and leak less
8071 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8073 * gst/rtsp-server/rtsp-media.c:
8074 media: don't leak the destination address
8076 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8078 * gst/rtsp-server/rtsp-client.c:
8079 * gst/rtsp-server/rtsp-media.c:
8080 * gst/rtsp-server/rtsp-media.h:
8081 * gst/rtsp-server/rtsp-session.c:
8082 * gst/rtsp-server/rtsp-session.h:
8083 rtsp: use RTCP to keep the session alive
8084 Use the RTCP rtcp-from stats field to find the associated session and use this
8085 to keep the session alive.
8087 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8089 * gst/rtsp-server/rtsp-session.c:
8090 session: add 5sec to the real session timeout
8091 Allow the session to live 5sec longer before really timing out. This should give
8092 clients some extra time to keep the session active.
8094 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-client.c:
8097 client: replay OK to GET/SET_PARAMETER
8098 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8099 so that we return OK for those requests.
8101 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8103 * gst/rtsp-server/rtsp-media.c:
8104 * gst/rtsp-server/rtsp-media.h:
8105 media: keep track of active transports
8106 Keep track of which transport is active to avoid closing the connection too
8108 Remove the destination transport also when going to NULL.
8109 Print some stats about the SDES and other RTCP messages we receive from the
8112 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8114 * examples/.gitignore:
8115 * examples/Makefile.am:
8116 * examples/test-sdp.c:
8117 example: add SDP relay example
8119 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8121 * gst/rtsp-server/rtsp-media.c:
8122 media: also count active TCP connections
8124 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8126 * gst/rtsp-server/rtsp-media-factory.c:
8127 * gst/rtsp-server/rtsp-media.c:
8128 * gst/rtsp-server/rtsp-media.h:
8129 rtsp: add support for dynamic elements
8130 Add support for dynamic elements.
8131 Don't set live pipelines back to paused.
8133 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8135 * gst/rtsp-server/rtsp-sdp.c:
8136 sdp: don't add encoding name when absent in caps
8138 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8140 * gst/rtsp-server/rtsp-client.c:
8141 client: warn when we can't do RTP-Info
8143 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8145 * gst/rtsp-server/rtsp-media-factory.c:
8146 factory: factor out the stream construction
8148 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8150 * gst/rtsp-server/rtsp-client.c:
8151 client: only add RTP-Info when we have the info
8152 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8155 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8160 === release 0.10.3 ===
8162 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8166 - Fixes a bug where it put the wrong verion in pkgconfig
8167 - Link RTP and RTCP sources
8169 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8171 * gst/rtsp-server/rtsp-media.c:
8172 * gst/rtsp-server/rtsp-media.h:
8173 media: link the RTP udpsrc to the session manager
8174 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8175 shut down when the client sends a packet to open firewalls.
8177 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8179 * pkgconfig/gst-rtsp-server.pc.in:
8180 Don't use hard-coded version number in pkg-config file
8182 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8187 === release 0.10.2 ===
8189 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8194 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8197 * common/m4/.gitignore:
8198 * examples/.gitignore:
8199 * pkgconfig/.gitignore:
8200 add some .gitignore files
8202 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8204 * gst/rtsp-server/rtsp-media.c:
8205 media: seek to key frames
8207 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8209 * gst/rtsp-server/rtsp-media.c:
8210 media: emit the unprepared signal by id
8211 Emit the unprepared signal by id instead of name and set the media as
8214 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8216 * gst/rtsp-server/rtsp-media.c:
8217 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8219 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8221 * gst/rtsp-server/rtsp-server.c:
8222 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8224 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8226 * bindings/vala/gst-rtsp-server-0.10.vapi:
8227 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8228 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8229 Updated vala bindings
8231 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8233 * gst/rtsp-server/Makefile.am:
8234 * gst/rtsp-server/rtsp-client.c:
8235 * gst/rtsp-server/rtsp-media.c:
8236 server: use appsink and appsrc with the API
8237 Use the appsink/appsrc API instead of the signals for higher
8240 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8242 * examples/test-ogg.c:
8243 tests: set the payload type correctly
8245 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8247 * gst/rtsp-server/rtsp-media-factory.c:
8248 factory: connect to the unprepare signal
8249 Connect to the unprepare signal for non-reusable media so that we can remove
8250 them from the cache.
8252 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8254 * gst/rtsp-server/rtsp-media.c:
8255 * gst/rtsp-server/rtsp-media.h:
8256 media: add signal to notify of unprepare
8258 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8260 * gst/rtsp-server/rtsp-media.c:
8261 * gst/rtsp-server/rtsp-media.h:
8262 media: more work on making the media shared
8263 Add a reusable flag to medias, indicating that they can be reused after a state
8267 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * examples/test-readme.c:
8270 examples: mark the example as shared for testing
8272 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8274 * gst/rtsp-server/rtsp-media.c:
8275 * gst/rtsp-server/rtsp-media.h:
8276 client: support shared media
8277 Always perform the state actions even if the target state of the pipeline is
8278 already correct, we still want to add/remove the transports when we are dealing
8280 Keep a counter of the number of active transports for a media so that we can use
8281 this to perform a state change when needed.
8282 Perform a state change of the pipeline only when the first transport was added
8283 or when there are no active transports.
8285 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8287 * gst/rtsp-server/rtsp-client.c:
8288 client: fix refcounting crasher
8289 Don't need to remove the weak refs in the finalize methods, they are already
8290 removed in the dispose.
8291 Don't register the callback with a DestroyNofity.
8293 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8295 * gst/rtsp-server/rtsp-client.c:
8296 Fix rtsp client refcount management in TCP mode.
8297 Don't unref a client ref we never had. Fixes an unref
8298 of an already-free client object after a client
8299 teardown request for me.
8301 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8303 * gst/rtsp-server/rtsp-session.c:
8304 docs: fix typo in API docs
8306 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8308 * gst/rtsp-server/rtsp-media.c:
8310 Keep the udp sources in playing even if we go to paused. unlock the sources when
8312 Add some more debug info.
8313 Only seek when we need to.
8314 Keep track of the position when we go to paused.
8316 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8318 * gst/rtsp-server/rtsp-client.c:
8319 * gst/rtsp-server/rtsp-media.c:
8320 * gst/rtsp-server/rtsp-media.h:
8321 Add beginnings of seeking.
8322 Parse the Range header and perform a seek on the pipeline for the requested
8323 position. It's disabled currently until I figure out what's going wrong.
8325 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8327 * gst/rtsp-server/rtsp-client.c:
8328 allow pause requests for now.
8331 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8333 * gst/rtsp-server/rtsp-client.c:
8334 Remove weak ref on the session in teardown
8335 We need to remove our weakref from the session when we do a teardown because
8336 else we close the TCP connection prematurely.
8338 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8340 * gst/rtsp-server/rtsp-client.c:
8341 * gst/rtsp-server/rtsp-client.h:
8342 * gst/rtsp-server/rtsp-session-pool.c:
8343 Do some more session cleanup
8344 Make session timeout kill the TCP connection that currently watches the
8346 Remove the client timeout property.
8348 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8350 * gst/rtsp-server/rtsp-client.c:
8351 * gst/rtsp-server/rtsp-client.h:
8352 * gst/rtsp-server/rtsp-media.c:
8353 * gst/rtsp-server/rtsp-media.h:
8354 * gst/rtsp-server/rtsp-server.c:
8355 * gst/rtsp-server/rtsp-session.c:
8356 * gst/rtsp-server/rtsp-session.h:
8358 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8361 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8363 * examples/Makefile.am:
8364 * examples/test-launch.c:
8365 Add example server that takes launch lines
8366 Add an example server that streams any -launch line.
8368 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8370 * examples/test-readme.c:
8371 * gst/rtsp-server/rtsp-client.c:
8372 * gst/rtsp-server/rtsp-media.c:
8373 * gst/rtsp-server/rtsp-media.h:
8374 Add support for live streams
8375 Add support for live streams and ranges
8376 Start on handling TCP data transfer.
8378 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8380 * gst/rtsp-server/rtsp-media.c:
8381 Free the pipeline before other things
8384 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8386 * gst/rtsp-server/rtsp-client.c:
8387 Only free the pending tunnel if there is one
8390 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8392 * gst/rtsp-server/rtsp-client.c:
8393 * gst/rtsp-server/rtsp-client.h:
8394 * gst/rtsp-server/rtsp-media.c:
8395 rtsp-server: Add support for tunneling
8396 Add support for tunneling over HTTP.
8397 Use new connection methods to retrieve the url.
8398 Dispatch messages based on the message type instead of blindly
8399 assuming it's always a request.
8400 Keep track of the watch id so that we can remove it later.
8401 Set the media pipeline to NULL before unreffing the pipeline.
8403 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8405 * gst/rtsp-server/rtsp-client.c:
8406 * gst/rtsp-server/rtsp-client.h:
8407 Fix for channel -> watch rename in gstreamer
8408 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8410 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8412 * gst/rtsp-server/rtsp-client.c:
8413 * gst/rtsp-server/rtsp-client.h:
8415 Use the async RTSP channels instead of spawning a new thread for each client.
8416 If a sessionid is specified in a request, fail if we don't have the session.
8418 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8420 * gst/rtsp-server/rtsp-media.c:
8421 Add better debug info
8422 Add some better debug info.
8424 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8426 * examples/test-video.c:
8428 Add support for session timeouts in the example.
8430 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8432 * gst/rtsp-server/rtsp-session-pool.c:
8433 * gst/rtsp-server/rtsp-session-pool.h:
8434 Pass GTimeVal around for performance reasons
8435 Get the current time only once and pass it around so that sessions don't have to
8436 get the current time anymore.
8437 Add experimental support for a GSource that dispatches when the session needs to
8440 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-session.c:
8443 * gst/rtsp-server/rtsp-session.h:
8444 Add better support for session timeouts
8445 Add a method to request the number of milliseconds when a session will timeout.
8447 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8449 * gst/rtsp-server/rtsp-media.c:
8450 * gst/rtsp-server/rtsp-media.h:
8451 Add suport for RTP manager monitoring
8452 Add the first stage in monitoring the rtp manager.
8453 Make sure we don't update the state to something we don't want.
8455 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8457 * gst/rtsp-server/rtsp-client.c:
8458 Add support for session keepalive
8459 Get and update the session timeout for all requests. get the session as early as
8462 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8464 * gst/rtsp-server/rtsp-media-factory.h:
8465 * gst/rtsp-server/rtsp-media.c:
8466 * gst/rtsp-server/rtsp-media.h:
8467 Handle media bus messages
8468 Handle media bus messages in a custom mainloop and dispatch them to the
8469 RTSPMedia objects. Let the default implementation handle some common messages.
8471 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8473 * gst/rtsp-server/rtsp-client.c:
8474 * gst/rtsp-server/rtsp-session-pool.c:
8475 * gst/rtsp-server/rtsp-session.c:
8476 Some more session timeout handling
8477 Move the session header setting code to a central place so that we always add
8478 the timeout parameter too.
8479 Handle timeouts by running the session cleanup code.
8480 Stop media before cleaning up.
8482 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/rtsp-client.c:
8485 * gst/rtsp-server/rtsp-client.h:
8486 Add timeout property
8487 Add a timeout property ot the client and make the other properties into GObject
8490 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8492 * gst/rtsp-server/rtsp-session-pool.c:
8493 Use getters and setters in property code
8494 Use the getters and setters for the timeout property instead of locking
8497 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8499 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8501 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8503 * gst/rtsp-server/rtsp-session-pool.c:
8504 * gst/rtsp-server/rtsp-session-pool.h:
8505 * gst/rtsp-server/rtsp-session.c:
8506 * gst/rtsp-server/rtsp-session.h:
8507 Add more timeout stuff
8508 Add method to check if a session is expired.
8509 Add method to perform cleanup on a session pool.
8511 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8513 * gst/rtsp-server/rtsp-client.c:
8514 * gst/rtsp-server/rtsp-session-pool.c:
8515 * gst/rtsp-server/rtsp-session-pool.h:
8516 * gst/rtsp-server/rtsp-session.c:
8517 * gst/rtsp-server/rtsp-session.h:
8518 Add beginnings of session timeouts and limits
8519 Add the timeout value to the Session header for unusual timeout values.
8520 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8521 limit on the amount of retry we do after a sessionid collision.
8522 Add properties to the sessionid and the timeout of a session. Keep track of
8523 creation time and last access time for sessions.
8525 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8527 * gst/rtsp-server/rtsp-client.c:
8528 * gst/rtsp-server/rtsp-media.c:
8529 * gst/rtsp-server/rtsp-media.h:
8530 * gst/rtsp-server/rtsp-sdp.c:
8531 * gst/rtsp-server/rtsp-session-pool.c:
8532 * gst/rtsp-server/rtsp-session.c:
8533 * gst/rtsp-server/rtsp-session.h:
8534 Cleanup of sessions and more
8535 Fix the refcounting of media and sessions in the client. Properly clean up the
8536 session data when the client performs a teardown.
8537 Add Server header to responses.
8538 Allow for multiple uri setups in one session.
8539 Add Range header to the PLAY response and add the range attribute to the SDP
8541 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8542 give the ownership of the sessionid to the session object.
8544 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/rtsp-server/rtsp-server.c:
8547 * gst/rtsp-server/rtsp-server.h:
8549 Rename the 'server_port' variable to simply 'port'.
8551 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8554 * gst/rtsp-server/rtsp-client.c:
8555 * gst/rtsp-server/rtsp-media.c:
8556 * gst/rtsp-server/rtsp-media.h:
8557 * gst/rtsp-server/rtsp-session.c:
8558 * gst/rtsp-server/rtsp-session.h:
8559 Rework the way we handle transports for streams
8560 Make the media accept an array of transports for the streams that we have
8561 configured for the play/pause requests.
8562 Implement server states for a client and its media.
8563 Require 0.10.22.1 (git HEAD) of gstreamer.
8565 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8567 * gst/rtsp-server/rtsp-client.c:
8568 * gst/rtsp-server/rtsp-media-factory.c:
8569 Drop const from functions dealing with urls
8570 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8571 have the right const in them.
8573 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8575 * gst/rtsp-server/rtsp-client.c:
8576 * gst/rtsp-server/rtsp-media.c:
8577 * gst/rtsp-server/rtsp-sdp.c:
8581 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8583 * gst/rtsp-server/rtsp-client.c:
8584 * gst/rtsp-server/rtsp-media-factory.c:
8585 * gst/rtsp-server/rtsp-media.c:
8586 * gst/rtsp-server/rtsp-media.h:
8588 Don't keep a reference to the GstRTSPMedia in the stream.
8589 Free more things when freeing the GstRTSPMedia.
8591 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8594 * gst/rtsp-server/rtsp-media-factory.c:
8595 * gst/rtsp-server/rtsp-media-factory.h:
8596 * gst/rtsp-server/rtsp-media.c:
8597 * gst/rtsp-server/rtsp-media.h:
8598 * gst/rtsp-server/rtsp-server.c:
8599 * gst/rtsp-server/rtsp-server.h:
8600 More docs and small cleanups
8601 Add some more docs and update the README
8602 Cleanup some method names.
8603 Remove an unneeded idx field in the GstRTSPMediaStream
8605 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8608 * examples/Makefile.am:
8609 * examples/test-readme.c:
8610 Add a README and more example code
8611 Add a README file that contains a small introduction on how to use the server
8612 along with the example code explained in the readme.
8614 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * gst/rtsp-server/rtsp-media.c:
8617 * gst/rtsp-server/rtsp-server.c:
8618 Fix some leaks and change default port
8619 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8620 we finished the initial preroll. If we keep them locked, setting the pipeline to
8621 NULL will not stop and clean up the sources correctly.
8622 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8624 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * gst/rtsp-server/rtsp-session.c:
8627 * gst/rtsp-server/rtsp-session.h:
8628 Cleanups to the session object
8629 Remove some unneeded variables in the session state of a stream such as the
8630 owner media and the server transport.
8631 Get the configuration of a media stream in a session based on the media_stream
8632 in the original object instead of our cached index.
8633 Free more data in the finalize method.
8635 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8637 * gst/rtsp-server/rtsp-client.c:
8638 * gst/rtsp-server/rtsp-client.h:
8639 Cleanups and reuse media from DESCRIBE
8640 Handle thread create errors.
8641 Rename some internal methods to better match what they actually do.
8642 Handle misconfiguration of session_pool and media_mapping gracefully.
8643 Cache the DESCRIBE media and uri in the client connection and reuse them when
8644 we receive a SETUP request in the same connection for the same uri.
8645 Cleanup the client connection object.
8647 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8649 * gst/rtsp-server/rtsp-media-factory.c:
8650 * gst/rtsp-server/rtsp-media-factory.h:
8651 * gst/rtsp-server/rtsp-media.c:
8652 * gst/rtsp-server/rtsp-media.h:
8653 Add shared properties to media and factory
8654 Add the shared property to media.
8655 Implement some simple caching in the factory depending on if the media is shared
8658 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8660 * gst/rtsp-server/rtsp-client.c:
8661 Add a little comment
8662 Add some comment about the content-base header.
8664 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8666 * examples/Makefile.am:
8668 * examples/test-mp4.c:
8669 * examples/test-ogg.c:
8670 * examples/test-video.c:
8671 * gst/rtsp-server/Makefile.am:
8672 * gst/rtsp-server/rtsp-client.c:
8673 * gst/rtsp-server/rtsp-client.h:
8674 * gst/rtsp-server/rtsp-media-factory.c:
8675 * gst/rtsp-server/rtsp-media-factory.h:
8676 * gst/rtsp-server/rtsp-media.c:
8677 * gst/rtsp-server/rtsp-media.h:
8678 * gst/rtsp-server/rtsp-sdp.c:
8679 * gst/rtsp-server/rtsp-sdp.h:
8680 * gst/rtsp-server/rtsp-server.c:
8681 * gst/rtsp-server/rtsp-server.h:
8682 * gst/rtsp-server/rtsp-session.c:
8683 * gst/rtsp-server/rtsp-session.h:
8684 Reorganize things, prepare for media sharing
8685 Added various other test server examples
8686 Move the SDP message generation to a separate helper.
8687 Refactor common code for finding the session.
8688 Add content-base for realplayer compatibility
8689 Clean up request uris before processing for better vlc compatibility.
8690 Move prerolling and pipeline construction to the RTSPMedia object.
8691 Use multiudpsink for future pipeline reuse.
8693 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8699 === release 0.10.1 ===
8701 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8707 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8709 * bindings/vala/Makefile.am:
8711 Add more directories and files to the dist.
8713 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8715 * bindings/python/Makefile.am:
8716 * bindings/python/rtspserver.override:
8717 Fixed compile error of python bindings
8719 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8721 * bindings/vala/gst-rtsp-server-0.10.vapi:
8722 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8723 Marked values as nullable accordingly
8725 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8727 * bindings/vala/gst-rtsp-server-0.10.vapi:
8728 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8729 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8730 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8731 Updated Vala bindings
8733 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8735 * gst/rtsp-server/rtsp-client.c:
8736 * gst/rtsp-server/rtsp-media-mapping.c:
8737 * gst/rtsp-server/rtsp-media-mapping.h:
8738 * gst/rtsp-server/rtsp-media.h:
8739 * gst/rtsp-server/rtsp-session-pool.h:
8740 Cleanups and doc updates
8741 Add some more documentation and do some minor cleanups here and there.
8743 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8745 * gst/rtsp-server/rtsp-client.c:
8746 * gst/rtsp-server/rtsp-media-factory.c:
8747 * gst/rtsp-server/rtsp-media-factory.h:
8748 * gst/rtsp-server/rtsp-media.c:
8749 * gst/rtsp-server/rtsp-media.h:
8750 * gst/rtsp-server/rtsp-session.c:
8751 * gst/rtsp-server/rtsp-session.h:
8753 Rename GstRTSPMediaBin to GstRTSPMedia
8754 Parse the request url into a GstRTSPUri object and pass this object to the
8755 various handlers and methods that require the uri.
8757 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8761 Add some more docs and remove some old code from the example.
8763 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8765 * gst/rtsp-server/rtsp-client.c:
8766 Handle state change failures better
8767 Handle state change failures better when changing the state of the pipeline to
8770 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8772 * gst/rtsp-server/rtsp-media-factory.c:
8773 * gst/rtsp-server/rtsp-media-factory.h:
8774 Make element creation more extendible
8775 Add get_element vmethod to the default MediaFactory so that subclasses can just
8776 override that method and still use the default logic for making a MediaBin from
8779 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8782 * gst/rtsp-server/Makefile.am:
8783 * gst/rtsp-server/rtsp-client.c:
8784 * gst/rtsp-server/rtsp-client.h:
8785 * gst/rtsp-server/rtsp-media-factory.c:
8786 * gst/rtsp-server/rtsp-media-factory.h:
8787 * gst/rtsp-server/rtsp-media-mapping.c:
8788 * gst/rtsp-server/rtsp-media-mapping.h:
8789 * gst/rtsp-server/rtsp-media.c:
8790 * gst/rtsp-server/rtsp-media.h:
8791 * gst/rtsp-server/rtsp-server.c:
8792 * gst/rtsp-server/rtsp-server.h:
8793 * gst/rtsp-server/rtsp-session.c:
8794 * gst/rtsp-server/rtsp-session.h:
8795 Make the server handle arbitrary pipelines
8796 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8797 The GstMediaBin object has a handle to a bin with elements and to a list of
8798 GstMediaStream objects that this bin produces.
8799 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8800 with methods to register and remove those mappings.
8801 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8802 used by the server instance.
8803 Modify the example application so that it shows how to create custom pipelines
8804 attached to a specific mount point.
8805 Various misc cleanps.
8807 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8809 * gst/rtsp-server/rtsp-server.c:
8810 * gst/rtsp-server/rtsp-server.h:
8811 Allow setting a custom media factory for a server
8813 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8815 * gst/rtsp-server/rtsp-client.c:
8816 * gst/rtsp-server/rtsp-client.h:
8817 Allow setting a custom media factory for a client.
8819 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8821 * gst/rtsp-server/Makefile.am:
8822 Add Makefile entry for the media factory
8824 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8826 * gst/rtsp-server/rtsp-media-factory.c:
8827 * gst/rtsp-server/rtsp-media-factory.h:
8828 Add media factory to map urls to media pipeline objects.
8830 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8832 * gst/rtsp-server/rtsp-media.c:
8833 * gst/rtsp-server/rtsp-media.h:
8834 Add comments. Remove unused field
8836 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8838 * gst/rtsp-server/rtsp-session-pool.c:
8839 * gst/rtsp-server/rtsp-session-pool.h:
8840 Allow custom session pools to override the session id allocation algorithms Add some comments.
8842 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-session.h:
8847 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8849 * gst/rtsp-server/rtsp-client.c:
8850 * gst/rtsp-server/rtsp-client.h:
8851 Move the connection code in one place Add some comments
8853 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8855 * gst/rtsp-server/rtsp-server.c:
8856 * gst/rtsp-server/rtsp-server.h:
8857 Make vmethod to create and accept new clients. Add some docs.
8859 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8861 * gst/rtsp-server/rtsp-server.c:
8862 * gst/rtsp-server/rtsp-server.h:
8863 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8865 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8867 * gst/rtsp-server/rtsp-client.c:
8868 * gst/rtsp-server/rtsp-client.h:
8869 Name the parameters more appropriately.
8871 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8873 * gst/rtsp-server/rtsp-session-pool.c:
8874 Do some more cleanup of the session pool.
8876 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8878 * gst/rtsp-server/Makefile.am:
8879 * gst/rtsp-server/rtsp-client.c:
8880 Check if return value of gst_rtsp_session_get_media is not NULL
8882 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8884 * gst/rtsp-server/Makefile.am:
8885 Install rtsp-session and rtsp-session-pool headers
8887 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8892 * bindings/python/Makefile.am:
8893 * bindings/python/arg-types.py:
8894 * bindings/python/codegen/Makefile.am:
8895 * bindings/python/codegen/__init__.py:
8896 * bindings/python/codegen/argtypes.py:
8897 * bindings/python/codegen/code-coverage.py:
8898 * bindings/python/codegen/codegen.py:
8899 * bindings/python/codegen/definitions.py:
8900 * bindings/python/codegen/defsparser.py:
8901 * bindings/python/codegen/docextract.py:
8902 * bindings/python/codegen/docgen.py:
8903 * bindings/python/codegen/fileprefix.override:
8904 * bindings/python/codegen/fileprefixmodule.c:
8905 * bindings/python/codegen/h2def.py:
8906 * bindings/python/codegen/mergedefs.py:
8907 * bindings/python/codegen/mkskel.py:
8908 * bindings/python/codegen/override.py:
8909 * bindings/python/codegen/reversewrapper.py:
8910 * bindings/python/codegen/scmexpr.py:
8911 * bindings/python/rtspserver-types.defs:
8912 * bindings/python/rtspserver.defs:
8913 * bindings/python/rtspserver.override:
8914 * bindings/python/rtspservermodule.c:
8916 Add python bindings.
8918 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8920 * bindings/Makefile.am:
8922 Don't go into python dir when requirements for python bindings are missing
8924 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8926 * bindings/Makefile.am:
8927 * bindings/vala/Makefile.am:
8929 Install Vala bindings if vala is available
8931 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8933 * bindings/vala/gst-rtsp-server-0.10.deps:
8934 * bindings/vala/gst-rtsp-server-0.10.vapi:
8935 * bindings/vala/gst-rtsp-server.vapi:
8936 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8937 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8938 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8939 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8940 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8941 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8942 * bindings/vala/packages/gst-rtsp-server.deps:
8943 * bindings/vala/packages/gst-rtsp-server.excludes:
8944 * bindings/vala/packages/gst-rtsp-server.files:
8945 * bindings/vala/packages/gst-rtsp-server.gi:
8946 * bindings/vala/packages/gst-rtsp-server.metadata:
8947 * bindings/vala/packages/gst-rtsp-server.namespace:
8948 Regenerated Vala bindings
8950 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8952 * bindings/vala/gst-rtsp-server.vapi:
8953 * bindings/vala/packages/gst-rtsp-server.metadata:
8954 Fixed typo in included headers for vala bindings
8956 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8960 * pkgconfig/Makefile.am:
8961 * pkgconfig/gst-rtsp-server.pc.in:
8962 Added pkgconfig file
8964 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8966 * bindings/vala/gst-rtsp-server.vapi:
8967 * bindings/vala/packages/gst-rtsp-server.excludes:
8968 * bindings/vala/packages/gst-rtsp-server.gi:
8969 * bindings/vala/packages/gst-rtsp-server.metadata:
8970 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8972 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8974 * bindings/vala/gst-rtsp-server.vapi:
8975 * bindings/vala/packages/gst-rtsp-server.deps:
8976 * bindings/vala/packages/gst-rtsp-server.files:
8977 * bindings/vala/packages/gst-rtsp-server.gi:
8978 * bindings/vala/packages/gst-rtsp-server.metadata:
8979 * bindings/vala/packages/gst-rtsp-server.namespace:
8982 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8984 * gst/rtsp-server/rtsp-session.c:
8985 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8987 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8989 * examples/Makefile.am:
8990 * gst/rtsp-server/Makefile.am:
8991 Put GStreamer version in library name
8993 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8995 * examples/Makefile.am:
8996 * gst/rtsp-server/Makefile.am:
8997 Fix some issues to pass distcheck
8999 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * gst/rtsp-server/rtsp-server.c:
9002 Added port property to GstRTSPServer class.
9004 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9009 * examples/Makefile.am:
9012 * gst/rtsp-server/Makefile.am:
9013 * gst/rtsp-server/rtsp-client.c:
9014 * gst/rtsp-server/rtsp-client.h:
9015 * gst/rtsp-server/rtsp-media.c:
9016 * gst/rtsp-server/rtsp-media.h:
9017 * gst/rtsp-server/rtsp-server.c:
9018 * gst/rtsp-server/rtsp-server.h:
9019 * gst/rtsp-server/rtsp-session-pool.c:
9020 * gst/rtsp-server/rtsp-session-pool.h:
9021 * gst/rtsp-server/rtsp-session.c:
9022 * gst/rtsp-server/rtsp-session.h:
9025 * src/rtsp-client.c:
9026 * src/rtsp-client.h:
9029 * src/rtsp-server.c:
9030 * src/rtsp-server.h:
9031 * src/rtsp-session-pool.c:
9032 * src/rtsp-session-pool.h:
9033 * src/rtsp-session.c:
9034 * src/rtsp-session.h:
9035 Split in library and example program
9037 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9039 * src/rtsp-client.h:
9040 Removed obsolete variable
9042 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9044 * src/rtsp-client.c:
9045 * src/rtsp-client.h:
9046 Removed pipeline variable GstRTSPClient, because it's only used in one function
9048 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9051 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9053 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9055 * src/rtsp-session.c:
9056 Initialize some more vars.
9058 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9060 * src/rtsp-session.c:
9061 Initialize variable to avoid compiler warning.
9063 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9066 Add a reasonable generic .gitignore