1 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
4 meson: require matching GStreamer dep versions for unstable development releases
5 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
6 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
8 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
10 * tests/check/meson.build:
11 meson: update for meson.build_root() and .build_source() deprecation
12 -> use meson.project_build_root() or .global_build_root() instead.
13 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
15 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
18 * tests/check/meson.build:
19 meson: update for dep.get_pkgconfig_variable() deprecation
20 ... in favour of dep.get_variable('foo', ..) which in some
21 cases allows for further cleanups in future since we can
22 extract variables from pkg-config dependencies as well as
23 internal dependencies using this mechanism.
24 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
26 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
28 * gst/rtsp-server/meson.build:
29 * gst/rtsp-sink/meson.build:
30 rtsp-server: define G_LOG_DOMAIN
32 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
34 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
37 meson: bump meson requirement to >= 0.59
38 For monorepo build and ugly/bad, for advanced feature
39 option API like get_option('xyz').required(..) which
40 we use in combination with the 'gpl' option.
41 For rest of modules for consistency (people will likely
42 use newer features based on the top-level requirement).
43 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
45 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
48 meson: Streamline the way we detect when to build documentation
49 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
51 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
54 * gst/rtsp-server/meson.build:
56 meson: List libraries and their corresponding gir definition
57 Introduces a `libraries` variable that contains all libraries in a
58 list with the following format:
63 'gir': [ {full gir definition in a dict } ]
68 It therefore refactors the way we build the gir so that we can reuse the
69 same information to build them against 'gstreamer-full' in gst-build
70 when linking statically
71 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
73 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
75 * gst/rtsp-server/meson.build:
76 meson: Mark files as files()
77 Making it more robust and future proof
78 And fix issues that it creates
79 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
81 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
83 * gst/rtsp-server/rtsp-media.c:
84 rtsp-media: Unprepare suspended medias too
85 Previously suspended medias immediately reached the UNPREPARED state
86 without going through the media's unprepare() vfunc. This didn't allow
87 the media subclass to do any additional cleanup, and for example the
88 shutdown-eos property of GstRTSPMedia was ignored.
89 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
91 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
93 * gst/rtsp-server/rtsp-media.c:
94 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
95 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
97 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
99 * gst/rtsp-server/rtsp-client.c:
100 * gst/rtsp-server/rtsp-session.c:
101 * gst/rtsp-server/rtsp-session.h:
102 rtsp-client: make sure sessmedia will not get freed while used
103 handle_*_request() functions were all retrieving the session media from
104 the session by calling gst_rtsp_session_get_media () which is a transfer-none
105 call. If a session timeout happens at that time, the session media may get freed
106 making the pointer invalid..
108 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
110 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
112 * gst/rtsp-server/rtsp-media.c:
113 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
114 Previously the status was only changed for other medias.
115 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
117 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
119 * gst/rtsp-server/rtsp-session.c:
120 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
121 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
122 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
124 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
127 doc: update IRC links to OFTC
128 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
130 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
132 * docs/gst_plugins_cache.json:
135 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
137 === release 1.19.2 ===
139 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
144 * docs/gst_plugins_cache.json:
145 * gst-rtsp-server.doap:
149 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
151 * gst/rtsp-server/rtsp-media.c:
152 * gst/rtsp-server/rtsp-stream.c:
153 * gst/rtsp-server/rtsp-stream.h:
154 * gst/rtsp-sink/gstrtspclientsink.c:
155 Protection against early RTCP packets.
156 When receiving RTCP packets early the funnel is not ready yet and
157 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
158 This causes the thread that handle RTCP packets to go to pause mode.
159 Since this thread is in pause mode there will be no further callbacks to
160 handle keep-alive for incoming RTCP packets. This will make the session
161 time out if the client is not using another keep-alive mechanism.
162 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
163 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
165 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
169 Update COPYING.LIB, COPYING files
170 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
172 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
174 * docs/gst_plugins_cache.json:
178 === release 1.19.1 ===
180 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
185 * docs/gst_plugins_cache.json:
186 * gst-rtsp-server.doap:
190 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
192 * gst/rtsp-server/rtsp-stream.c:
193 rtsp-stream: use new gst_buffer_new_memdup()
194 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
196 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
198 * gst/rtsp-server/rtsp-media-factory-uri.c:
199 rtsp-media: fix leak when adding converter
200 Free the previous caps before reusing the variable for the converter caps.
201 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
203 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
205 * gst/rtsp-server/rtsp-client.c:
206 rtsp-client: fix leak adding headers
207 gst_rtsp_message_add_header() makes a copy of the header, instead
209 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
211 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
213 * gst/rtsp-server/rtsp-stream.c:
214 Use gst_element_request_pad_simple...
215 Instead of the deprecated gst_element_get_request_pad.
216 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
218 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
220 * gst/rtsp-server/rtsp-media.c:
221 rtsp-media: Ensure the bus watch is removed during unprepare
222 It's possible for the destruction of the source to be delayed.
223 Instead of relying on the dispose() to remove the bus watch, do
225 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
227 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
230 docs: minor spelling correction in README
231 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
233 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
235 * examples/test-replay-server.c:
236 test-replay-server: minor spelling corrections
237 Bumped on these while investigating the example code.
238 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
240 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
242 * tests/check/gst/stream.c:
243 tests: Don't fail tests if IPv6 not available.
244 On computers with IPv6 disabled it shouldn't result in a test failure.
245 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
247 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
249 * gst/rtsp-server/rtsp-media.c:
250 rtsp-media: Add one more case to seek avoidance
251 This is an extension to the previous commit. There can also be cases where the
252 start position is not specified, in those cases we should also avoid doing
253 seeking unless it's forced.
254 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
256 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
258 * gst/rtsp-server/rtsp-media.c:
259 rtsp-media: Improve skipping trickmode seek.
260 We can also skip the seek if the end range is already
262 Avoids initial seek on play start if playing full stream.
263 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
265 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
267 * gst/rtsp-sink/gstrtspclientsink.c:
268 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
269 It's sufficient to run them during the FIRST stage instead of in both.
270 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
272 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
274 * tests/check/gst/rtspclientsink.c:
275 tests: rtspclientsink: fix some leaks
276 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
278 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
280 * gst/rtsp-sink/gstrtspclientsink.c:
281 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
282 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
284 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
286 * tests/check/gst/rtspclientsink.c:
287 rtspclientsink: add unit test for potential shutdown deadlock
288 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
290 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
292 * gst/rtsp-sink/gstrtspclientsink.c:
293 rtspclientsink: fix deadlock on shutdown before preroll
294 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
295 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
297 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
299 * gst/rtsp-server/rtsp-stream.c:
300 rtsp-stream: avoid deadlock in send_func
301 Currently the send_func() runs in a thread of its own which is started
302 the first time we enter handle_new_sample(). It runs in an outer loop
303 until priv->continue_sending is FALSE, which happens when a TEARDOWN
304 request is received. We use a local variable, cont, which is initialized
305 to TRUE, meaning that we will always enter the outer loop, and at the
306 end of the outer loop we assign it the value of priv->continue_sending.
307 Within the outer loop there is an inner loop, where we wait to be
308 signaled when there is more data to send. The inner loop is exited when
309 priv->send_cookie has changed value, which it does when more data is
310 available or when a TEARDOWN has been received.
311 But if we get a TEARDOWN before send_func() is entered we will get stuck
312 in the inner loop because no one will increase priv->session_cookie
314 By not entering the outer loop in send_func() if priv->continue_sending
315 is FALSE we make sure that we do not get stuck in send_func()'s inner
316 loop should we receive a TEARDOWN before the send thread has started.
317 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
318 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
320 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
322 * gst/rtsp-server/rtsp-client.c:
323 rtsp-client: cleanup transports during TEARDOWN
324 When tunneling RTP over RTSP the stream transports are stored in a hash
325 table in the GstRTSPClientPrivate struct. They are used for, among other
326 things, mapping channel id to stream transports when receiving data from
327 the client. The stream tranports are created and added to the hash table
328 in handle_setup_request(), but unfortuately they are not removed in
329 handle_teardown_request(). This means that if the client sends data on
330 the RTSP connection after it has sent the TEARDOWN, which is often the
331 case when audio backchannel is enabled, handle_data() will still be able
332 to map the channel to a session transport and pass the data along to it.
333 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
334 because the stream is no longer joined to a bin.
335 We avoid this by removing the stream transports from the hash table when
336 we handle the TEARDOWN request.
337 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
339 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
341 * docs/gst_plugins_cache.json:
342 * gst/rtsp-sink/gstrtspclientsink.c:
343 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
344 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
346 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
348 * tests/check/gst/client.c:
349 Add test cases for mountpoint of '/'
350 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
352 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
354 * gst/rtsp-server/rtsp-client.c:
355 * gst/rtsp-server/rtsp-mount-points.c:
356 * gst/rtsp-server/rtsp-session-media.c:
357 Make a mount point of "/" work correctly.
358 As far as I can tell, this is neither explicitly allowed nor
359 forbidden by RFC 7826.
360 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
361 use in the wild (presumably with non-GStreamer servers).
362 GStreamer's prior behavior was confusing, in that
363 gst_rtsp_mount_points_add_factory() would appear to accept a mount
364 path of "" or "/", but later connection attempts would fail with a
365 "media not found" error.
366 This commit makes a mount path of "/" work for either form of URL,
367 while an empty mount path ("") is rejected and logs a warning.
368 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
370 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
372 * docs/gst_plugins_cache.json:
373 * gst/rtsp-sink/gstrtspclientsink.c:
374 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
375 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
377 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
379 * gst/rtsp-server/rtsp-media.c:
380 rtsp-media: Only count senders when counting blocked streams
381 Only sender streams sends the GstRTSPStreamBlocking message, so only
382 these should be counted before setting media status to prepared.
383 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
385 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
387 * gst/rtsp-sink/gstrtspclientsink.c:
388 rtspclientsink add proper support for uri queries
389 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
391 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
393 * gst/rtsp-server/rtsp-client.c:
394 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
395 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
396 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
398 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
400 * gst/rtsp-server/rtsp-stream.c:
401 rtsp-stream: collect a clock_rate when blocking
402 This lets us provide a clock_rate in a fashion similar to the
403 other code paths in get_rtpinfo()
404 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
406 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
408 * gst/rtsp-server/rtsp-media.c:
409 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
410 Otherwise this will cause memory corruption as the property expects a 64
412 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
414 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
416 * gst/rtsp-server/rtsp-media.c:
417 * gst/rtsp-server/rtsp-stream.c:
418 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
419 To prevent cases with prerolling when the inactive stream prerolls first
420 and the server proceeds without waiting for the active stream, we will
421 ignore GstRTSPStreamBlocking messages from incomplete streams. When
422 there are no complete streams (during DESCRIBE), we will listen to all
424 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
426 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
428 * tests/check/gst/media.c:
429 * tests/check/meson.build:
430 * tests/files/test.avi:
431 media test: Add test for seeking one active stream with a demuxer
432 Add another seek_one_active_stream test but with a demuxer. The demuxer
433 will flush both streams in opposed to the existing test which only
434 flushes the active stream. This will help exposing problems with the
435 prerolling process after a flushing seek.
436 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
438 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
440 * gst/rtsp-server/meson.build:
442 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
443 * pkgconfig/gstreamer-rtsp-server.pc.in:
444 * pkgconfig/meson.build:
445 Meson: Use pkg-config generator
446 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
448 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
451 meson: update glib minimum version to 2.56
452 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
454 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
456 * examples/test-launch.c:
457 * gst/rtsp-server/rtsp-media-factory.c:
458 * gst/rtsp-server/rtsp-media-factory.h:
459 * gst/rtsp-server/rtsp-media.c:
460 * gst/rtsp-server/rtsp-server-internal.h:
461 * gst/rtsp-server/rtsp-stream.c:
462 * tests/check/gst/client.c:
463 rtsp-media-factory: expose API to disable RTCP
464 This is supported by the RFC, and can be useful on systems where
465 allocating two consecutive ports is problematic, and RTCP is not
467 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
469 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
471 * hooks/pre-commit.hook:
473 git: use our standard pre commit hook
474 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
476 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
478 * gst/rtsp-server/rtsp-stream.c:
479 rtsp-stream: make use of blocked_running_time in query_position
480 When blocking, the sink element will not have received a buffer
481 yet and the position query will fail. Instead, we make use of
482 the running time of the buffer we blocked on.
483 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
485 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
487 * gst/rtsp-server/rtsp-stream.c:
488 rtsp-stream: collect rtp info when blocking
489 We don't unblock the stream anymore before replying to the
490 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
491 so the sinks don't have a last-sample after potentially flush
492 seeking. seek_trickmode waits for preroll however, which means
493 the stream will block and wait for a first buffer. Subsequent
494 calls to get_rtpinfo() can thus make use of the information.
495 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
496 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
498 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
500 * examples/meson.build:
501 * examples/test-replay-server.c:
502 * examples/test-replay-server.h:
503 examples: Add an example for loop playback
504 This demo example shows a way of file loop playback of a given source.
505 Note that client seek request is not properly implemented yet.
506 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
508 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
510 * gst/rtsp-server/rtsp-media.c:
511 rtsp-media: Plug memory leak
512 The get-storage signal of rtpbin increases the ref count of the storage.
513 So we have to unref it after usage.
514 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
516 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
518 * gst/rtsp-server/rtsp-media.c:
519 rtsp-media: Get rates only on sender streams
520 When play a media with both sender and receiver stream, like ONVIF
521 back channel audio in, gst_rtsp_media_get_rates call
522 gst_rtsp_stream_get_rates for each stream to set the rates. But
523 gst_rtsp_stream_get_rates return false for the receiver steam, which
524 lead a g_assert crash.
525 Instead to get rates on all streams, now just get rates on sender
527 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
529 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
531 * gst/rtsp-server/rtsp-media.c:
532 * gst/rtsp-server/rtsp-server-internal.h:
533 * gst/rtsp-server/rtsp-stream.c:
534 rtsp-media: set a 0 storage size for TCP receivers
535 ulpfec correction is obviously useless when receiving a stream
536 over TCP, and in TCP modes the rtp storage receives non
537 timestamped buffers, causing it to queue buffers indefinitely,
538 until the queue grows so large that sanity checks kick in and
539 warnings start to get emitted.
540 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
542 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
544 * gst/rtsp-server/rtsp-stream.c:
545 rtsp-stream: preroll on gap events
546 This allows negotiating a SDP with all streams present, but only
547 start sending packets at some later point in time
548 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
550 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
552 * gst/rtsp-server/rtsp-media.c:
553 rtsp-media: do not unblock on unsuspend
554 rtsp_media_unsuspend() is called from handle_play_request()
555 before sending the play response. Unblocking the streams here
556 was causing data to be sent out before the client was ready
557 to handle it, with obvious side effects such as initial packets
558 getting discarded, causing decoding errors.
559 Instead we can simply let the media streams be unblocked when
560 the state of the media is set to PLAYING, which occurs after
561 sending the play response.
562 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
564 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
567 ci: include template from gst-ci master branch again
569 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
571 * docs/gst_plugins_cache.json:
575 === release 1.18.0 ===
577 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
583 * docs/gst_plugins_cache.json:
584 * gst-rtsp-server.doap:
588 === release 1.17.90 ===
590 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
595 * docs/gst_plugins_cache.json:
596 * gst-rtsp-server.doap:
600 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
602 * gst/rtsp-server/rtsp-thread-pool.c:
603 rtsp-thread-pool.c: fix clang 10 warning
604 clang 10 is complaining about incompatible types due to the
607 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
609 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
611 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
613 * gst/rtsp-server/rtsp-thread-pool.c:
614 rtsp-thread-pool.c: fix clang 10 warning
615 clang 10 is complaining about incompatible types due to the
618 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
620 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
622 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
624 * gst/rtsp-server/rtsp-sdp.c:
625 rtsp-sdp: Fix resource leak in mikey messsage
626 Fixed a resource leak for mikey message while adding crypto session
628 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
630 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
633 * scripts/extract-release-date-from-doap-file.py:
634 meson: set release date from .doap file for releases
635 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
637 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
639 * gst/rtsp-server/rtsp-stream.c:
640 rtsp-stream: explicitly set caps on udpsrc elements
641 This causes them to send caps events before data flow, which is
642 usually a pretty correct thing to do!
643 Not doing so manifested in a bug where ssrcdemux wouldn't forward
644 the caps it had received with an extra ssrc field, as it hadn't
645 received any caps event.
647 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
649 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
651 * docs/gst_plugins_cache.json:
655 === release 1.17.2 ===
657 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
662 * docs/gst_plugins_cache.json:
663 * gst-rtsp-server.doap:
667 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
669 * docs/gst_plugins_cache.json:
670 doc: Stop documenting properties from parents
672 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
674 * docs/gst_plugins_cache.json:
675 docs: Fix version in the plugins cache
676 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
678 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
680 * gst/rtsp-sink/gstrtspclientsink.c:
681 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
682 It's deprecated, unneeded and doesn't do anything anymore.
683 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
685 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
690 === release 1.17.1 ===
692 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
697 * docs/gst_plugins_cache.json:
698 * gst-rtsp-server.doap:
702 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
704 * gst/rtsp-server/rtsp-media.c:
705 rtsp-media: Add/configure transports when completing the pipeline
706 Otherwise the transports are not set up yet during the PLAY request
707 handling when unsuspending (and thus unblocking) the media.
708 In case of live pipelines this then causes the first few packets to go
709 to the sinks before they know what to do with them, and they simply
710 discard them which is rather suboptimal in case of keyframes.
711 For non-live pipelines this is not a problem because the sink will still
712 be PAUSED and as such not send out the data yet but wait until it goes
713 to PLAYING, which is late enough.
714 Adding the transports multiple times is not a problem: if the transport
715 is already added it won't be added another time and TRUE will be
717 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
719 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
720 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
722 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
724 * gst/rtsp-server/rtsp-media.c:
725 rtsp-media: Fix misleading comment
726 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
728 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
730 * gst/rtsp-server/rtsp-media.c:
731 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
732 The pad probes are not needed anymore at this point and later when
733 reaching buffering 100% only the state is changed, no unblocking
735 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
737 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
739 * gst/rtsp-server/rtsp-media.c:
740 rtsp-media: Remove duplicated media_unblock() function
741 It does literally the same as media_streams_set_blocked(FALSE).
742 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
744 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
746 * examples/test-onvif-server.c:
747 test-onvif-server: cast ntp-offset property value to 64 bit
748 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
750 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
752 * docs/gst_plugins_cache.json:
753 docs: Update plugins cache
755 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
757 * examples/test-onvif-server.c:
758 * examples/test-onvif-server.h:
759 * gst/rtsp-server/rtsp-onvif-media-factory.h:
760 onvif-media-factory: define autoptr cleanup function
761 And have the factory in the onvif-server example inherit from
762 GstRTSPOnvifMediaFactory.
763 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
765 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
767 * docs/gst_plugins_cache.json:
768 docs: Update plugins cache
770 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
772 * tests/check/gst/rtspserver.c:
773 tests: enforce I420 format
774 Test was not enforcing a video format on videotestsrc. I420 was picked as it
775 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
776 true (gst-plugins-base!689).
777 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
779 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
781 * gst/rtsp-sink/gstrtspclientsink.c:
782 plugins: uddate gst_type_mark_as_plugin_api() calls
784 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
787 doc: Require hotdoc >= 0.11.0
789 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
791 * docs/gst_plugins_cache.json:
792 docs: Update gst_plugins_cache.json
794 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
796 * gst/rtsp-sink/gstrtspclientsink.c:
797 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
799 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
801 * gst/rtsp-server/meson.build:
802 meson: gir: remove bogus sources_top_dir kwarg
803 Doesn't actually exist. Was fixed differently in Meson
804 so that the user doesn't have to specify it.
805 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
807 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
809 * tests/check/meson.build:
810 tests: put registry into tests/check not the gst/ subdir
811 Underscorify the test name before setting GST_REGISTRY,
812 so the registry actually ends up in the current build dir
814 For consistency with the other modules, but should also
815 avoid problems on windows.
816 Also fix indentation of environment block.
817 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
819 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
821 * tests/check/meson.build:
822 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
823 If core is built as a subproject (e.g. as in gst-build), make sure to use
824 the gst-plugin-scanner from the built subproject. Without this, gstreamer
825 might accidentally use the gst-plugin-scanner from the install prefix if
826 that exists, which in turn might drag in gst library versions we didn't
827 mean to drag in. Those gst library versions might then be older than
828 what our current build needs, and might cause our newly-built plugins
829 to get blacklisted in the test registry because they rely on a symbol
830 that the wrongly-pulled in gst lib doesn't have.
831 This should fix running of unit tests in gst-build when invoking
832 meson test or ninja test from outside the devenv for the case where
833 there is an older or different-version gst-plugin-scanner installed
834 in the install prefix.
835 In case no gst-plugin-scanner is installed in the install prefix, this
836 will fix "GStreamer-WARNING: External plugin loader failed. This most
837 likely means that the plugin loader helper binary was not found or
838 could not be run. You might need to set the GST_PLUGIN_SCANNER
839 environment variable if your setup is unusual." warnings when running
841 In the case where we find GStreamer core via pkg-config we use
842 a newly-added pkg-config var "pluginscannerdir" to get the right
843 directory. This has the benefit of working transparently for both
844 installed and uninstalled pkg-config files/setups.
845 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
847 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
849 * tests/check/meson.build:
850 tests: gst-plugins-base and -bad plugins are required for the unit tests
851 Make hard requirement until we have more fine-grained control
852 in the unit tests. Of course the presence of the .pc file doesn't
853 imply that the plugins we need are actually there, but it's at
854 least a step in the right direction.
855 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
857 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
859 * tests/check/meson.build:
860 tests: pick up rtsp-server plugins from build directory only
861 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
863 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
865 * gst/rtsp-server/rtsp-media.c:
866 rtsp-media: wait for all GstRTSPStreamBlocking messages
867 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
868 each active stream when checking if all streams are blocked.
869 Without this change there will be a race condition when using two or
870 more streams and rtsp-media receives a GstRTSPStreamBlocking message
871 from one of the streams. This is because rtsp-media then checks if all
872 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
873 stream. This function call returns TRUE if the stream has sent a
874 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
875 receive this message. This would then result in that rtsp-media
876 erroneously thinks it is blocking all streams which could result in
877 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
878 preroll, this could result in that rtsp-media thinks that the pipeline
879 is prerolled even though that might not be the case.
880 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
882 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
884 * gst/rtsp-server/rtsp-media.c:
885 rtsp-media: update expected_async_done during suspend
886 Set expected_async_done to FALSE in default_suspend() if a state change
887 occurs and the return value from set_target_state() is something other
888 than GST_STATE_CHANGE_ASYNC.
889 Without this change there is a risk that expected_async_done will be
890 TRUE even though no asynchronous state change is taking place. This
891 could happen if the pipeline is set to PAUSED using
892 media_set_pipeline_state_locked(), an asynchronous state change starts
893 and then the media is suspended (which could result in a state change,
894 aborting the asynchronous state change). If the media is suspended
895 before the asynchronous state change ends then expected_async_done will
896 be TRUE but no asynchronous state change is taking place.
897 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
899 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
901 * gst/rtsp-server/rtsp-client.c:
902 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
903 There was a race condition where client was being finalized and
904 concurrently in some other thread the rtsp ctrl timout was relying on
905 client data that was being freed.
906 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
907 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
909 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
911 * gst/rtsp-server/rtsp-media-factory.c:
912 * gst/rtsp-server/rtsp-media-factory.h:
913 * gst/rtsp-server/rtsp-media.c:
914 * gst/rtsp-server/rtsp-media.h:
915 media-factory: complete DSCP QoS setting support
916 add dscp_qos setting support at factory and media level to setup IP DSCP
917 field of bounded UDP sinks.
918 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
919 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
921 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
923 * gst/rtsp-server/rtsp-client.c:
924 rtsp-client: Fix some race conditions around timeout source removal
925 We always need to take the lock while accessing it as otherwise another
926 thread might've removed it in the meantime. Also when destroying and
927 creating a new one, ensure that the mutex is not shortly unlocked in
928 between as during that time another one might potentially be created
930 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
932 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
934 * gst/rtsp-server/rtsp-media.c:
935 * gst/rtsp-server/rtsp-stream.c:
936 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
937 And the same for gst_rtsp_stream_get_rates().
938 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
940 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
942 * examples/test-onvif-server.c:
943 examples: test-onvif-server: fix compiler warnings on raspbian
944 Fix printf format for 64-bit variables.
945 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
947 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
949 * gst/rtsp-server/rtsp-stream-transport.c:
950 * gst/rtsp-server/rtsp-stream-transport.h:
951 * gst/rtsp-server/rtsp-stream.c:
952 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
953 The old API is preserved now and new API was added that provides the
954 additional parameter to the callback.
955 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
956 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
958 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
960 * gst/rtsp-server/rtsp-client.c:
961 rtsp-client: Store the timeout source by pointer instead of id
962 That way we don't have to retrieve it again from the main context when
963 destroying it but can directly do so.
964 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
966 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
968 * gst/rtsp-server/rtsp-client.c:
969 rtsp-client: Clean up watch/watch context and related state consistently
970 And assert that it was cleaned up properly before the client is
971 finalized. If something is still around when the client is shut down
972 then something went very wrong before.
973 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
975 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
977 * gst/rtsp-server/rtsp-client.c:
978 * tests/check/gst/rtspserver.c:
979 rtsp-client: Combine the pre-session and post-session timeout
980 They previously used the same state but different mechanisms and
981 functions, which was difficult to follow, error prone and simply
983 Also adjust the test for the post-session timeout a bit to be less racy
984 now that the timing has slightly changed.
985 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
987 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
989 * gst/rtsp-server/rtsp-client.c:
990 rtsp-client: Don't ever close the client connection directly when a session is torn down
991 There might be other sessions that are running over the same RTSP
992 connection and we should not simply close the client directly if one of
994 By default the connection will be closed once the client closes it or
995 the OS does. This behaviour can be adjusted with the
996 post-session-timeout property, which allows to close it automatically
997 from the server side after all sessions are gone and the given timeout
999 This reverts the previous commit.
1000 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1002 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1004 * gst/rtsp-server/rtsp-client.c:
1005 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1006 Instead of closing it never at all. Previously there was only code that
1007 closed the client asynchronously if sending the response happened
1008 asynchrously at a later time.
1009 Thanks to Christian M for debugging this issue.
1010 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1011 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1013 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1015 * gst/rtsp-server/rtsp-stream.c:
1016 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1017 Otherwise no sink is found for multicast sreams and the less accurate
1018 fallback is used to determine the current sequence number and timestamp.
1020 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1022 * gst/rtsp-server/rtsp-auth.c:
1023 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1024 When using the basic authentication scheme, we wouldn't validate that
1025 the authorization field of the credentials is not NULL and pass it on
1026 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1027 dereference the NULL pointer and crash.
1028 A specially crafted (read: invalid) RTSP header can cause this to
1030 As a solution, check for the authorization to be not NULL before
1031 continuing processing it and if it is simply fail authentication.
1032 This fixes CVE-2020-6095 and TALOS-2020-1018.
1033 Discovered by Peter Wang of Cisco ASIG.
1035 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1037 * gst/rtsp-server/rtsp-client.c:
1038 rtsp-client: Use watch_context before unref
1039 Move the usage of priv->watch_context to beginning of function
1040 gst_rtsp_client_finalize. Instead of use it after
1041 g_main_context_unref (priv->watch_context).
1043 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1045 * gst/rtsp-server/rtsp-stream.c:
1046 rtsp-stream: fix deadlock on transport removal
1047 We cannot take the RTSPStream lock while holding a transport backlog
1048 lock, as remove_transport may be called externally, which will
1049 take first the RTSPStream lock then the transport backlog lock.
1051 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1053 * gst/rtsp-server/rtsp-server-internal.h:
1054 * gst/rtsp-server/rtsp-stream-transport.c:
1055 * gst/rtsp-server/rtsp-stream.c:
1056 rtsp-stream: clear backlog when removing transport
1057 This ensures we don't end up calling any of transports' callbacks
1058 with a potentially unreffed user_data (in practice, a client that
1059 may have been removed)
1061 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1063 * gst/rtsp-server/rtsp-stream.c:
1064 rtsp-stream: marshal calls to send_tcp_message to a single thread
1065 In order to address the race condition pointed out at
1066 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1067 we get rid of the send thread pool, and instead spawn and manage
1068 a single thread to pull samples from app sinks and add them to
1069 the transport's backlogs.
1070 Additionally, we now also always go through the backlogs in order
1071 to simplify the logic.
1073 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1075 * gst/rtsp-server/rtsp-server-internal.h:
1076 * gst/rtsp-server/rtsp-stream-transport.c:
1077 * gst/rtsp-server/rtsp-stream.c:
1078 rtsp-stream: properly protect TCP backlog access
1080 We cannot hold stream->lock while pushing data, but need
1081 to consistently check the state of the backlog both from
1082 the send_tcp_message function and the on_message_sent function,
1083 which may or may not be called from the same thread.
1084 This commit introduces internal API to allow for potentially
1085 recursive locking of transport streams, addressing a race
1086 condition where the RTSP stream could push items out of order
1087 when popping them from the backlog.
1089 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1091 * gst/rtsp-server/rtsp-media.c:
1092 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1093 It's taken ownership of by the media, and returned with `transfer none`
1094 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1095 first then any bindings will wrongly take ownership of the pipeline once
1096 it arrives in bindings code.
1098 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1100 * examples/test-onvif-client.c:
1101 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1103 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1105 * gst/rtsp-server/rtsp-media.c:
1106 rtsp-media: fix default latency
1108 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1110 * gst/rtsp-server/rtsp-client.c:
1111 rtsp-client: make closing more thread safe
1112 + Take the watch lock prior to using priv->watch
1113 + Flush both the watch and connection before closing / unreffing
1114 gst_rtsp_connection_close() is not threadsafe on its own, this is
1115 a workaround at the client level, where we control both the watch
1118 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1120 * gst/rtsp-server/rtsp-latency-bin.c:
1121 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1124 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1125 `your_type_get_instance_private()` function instead
1128 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1130 * gst/rtsp-server/rtsp-client.c:
1131 * tests/check/gst/rtspserver.c:
1132 rtsp-client: add property post-session-timeout
1133 This is a TCP connection timeout for client connections, in seconds.
1134 If a positive value is set for this property, the client connection
1135 will be kept alive for this amount of seconds after the last session
1136 timeout. For negative values of this property the connection timeout
1137 handling is delegated to the system (just as it was before).
1140 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1142 * gst/rtsp-server/rtsp-stream.c:
1143 rtsp-stream: check for NULL transports prior to ref'ing
1145 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1147 * gst/rtsp-server/rtsp-server-internal.h:
1148 * gst/rtsp-server/rtsp-stream-transport.c:
1149 * gst/rtsp-server/rtsp-stream.c:
1150 rtsp-stream: fix checking of TCP backpressure
1151 The internal index of our appsinks, while it can be used to
1152 determine whether a message is RTP or RTCP, is not necessarily
1153 the same as the interleaved channel. Let the stream-transport
1154 determine the channel to check backpressure for, the same way
1155 it determines the channel according to whether it is sending
1158 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1160 * gst/rtsp-server/rtsp-session.c:
1161 rtsp-session: Butcher the file to please gst-indent in the CI
1162 This should be reverted once the CI has an updated gst-indent.
1164 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1166 * gst/rtsp-server/rtsp-session.c:
1167 * gst/rtsp-server/rtsp-session.h:
1168 * gst/rtsp-sink/gstrtspclientsink.c:
1169 * gst/rtsp-sink/gstrtspclientsink.h:
1170 rtsp-session & client: Remove deprecated GTimeVal
1171 GTimeVal won't work past 2038
1173 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1175 * gst/rtsp-server/rtsp-auth.c:
1176 rtsp-auth: fix default token leak
1178 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1180 * gst/rtsp-sink/gstrtspclientsink.c:
1181 gstrtspclientsink: unref transports when closing bin
1184 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1186 * gst/rtsp-server/rtsp-media.c:
1187 rtsp-media: Force seek when flush flag is set
1188 The commit "rtsp-client: define all seek accuracy flags from
1189 setup_play_mode" changed the behaviour of when doing a seek.
1190 Before that commit, having the flush flag set would result in a seek
1192 Even if no seek was needed. One reason to force seek is to flush old buffers
1193 created in Describe requests.
1194 Thus adding force seek also for flush flag will result in play request
1197 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1199 * gst/rtsp-server/rtsp-client.c:
1200 rtsp-client: Revitalize dead code
1201 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1204 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1206 * gst/rtsp-server/rtsp-sdp.c:
1207 rtsp-sdp: Don't try to use non-initialized values
1208 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1209 returns TRUE. Also avoid the whole clock signalling block if we're not
1210 dealing with senders.
1215 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1217 * gst/rtsp-server/rtsp-stream-transport.c:
1218 * gst/rtsp-server/rtsp-stream.c:
1219 * tests/check/gst/stream.c:
1220 rtsp-stream: Removing invalid transports returns false
1221 When removing transports an assertion was that the transports passed in
1222 for removal are present in the list, however that can't be assumed.
1223 As an example if a transport was removed from a thread running
1224 send_tcp_message, the main thread can try to remove the same transport
1225 again if it gets a handle_pause_request. This will not effect the
1226 transport list but it will effect n_tcp_transports as it will be
1227 decrement and then have the wrong value.
1229 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1231 * tests/check/gst/client.c:
1232 client test: add scale and speed negative tests
1233 Negative tests for scale and speed should be done as well, verify that
1234 the response code is "400 Bad request" when a bad request is done.
1236 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1238 * gst/rtsp-server/rtsp-auth.c:
1239 * gst/rtsp-server/rtsp-client.c:
1240 * gst/rtsp-server/rtsp-media-factory.c:
1241 * gst/rtsp-server/rtsp-media.c:
1242 * gst/rtsp-server/rtsp-server.c:
1243 * gst/rtsp-server/rtsp-session-pool.c:
1244 * gst/rtsp-server/rtsp-stream.c:
1245 * gst/rtsp-sink/gstrtspclientsink.c:
1246 Don't pass default GLib marshallers for signals
1247 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1248 actually internally optimize the signal (if the marshaller is available
1249 in GLib itself) by also setting the valist marshaller. This makes the
1250 signal emission a bit more performant than the regular marshalling,
1251 which still needs to box into `GValue` and call libffi in case of a
1253 Note that for custom marshallers, one would use
1254 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1256 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1258 * gst/rtsp-server/rtsp-mount-points.c:
1259 GstRTSPMountPoints: Remove any existing factory before adding a new one
1260 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1261 previous mount point will be freed" which was true when it was
1262 implemented using a GHashTable. But in 2012 it got rewrote using a
1263 GSequence and since then it could have 2 factories for the same path.
1264 Which one gets used is random, depending on the sorting order of 2
1267 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1269 * gst/rtsp-server/rtsp-client.c:
1270 * gst/rtsp-server/rtsp-server-internal.h:
1271 * gst/rtsp-server/rtsp-stream-transport.c:
1272 * gst/rtsp-server/rtsp-stream-transport.h:
1273 * gst/rtsp-server/rtsp-stream.c:
1274 stream: refactor TCP backpressure handling
1275 The previous implementation stopped sending TCP messages to
1276 all clients when a single one stopped consuming them, which
1277 obviously created problems for shared media.
1278 Instead, we now manage a backlog in stream-transport, and slow
1279 clients are removed once this backlog exceeds a maximum duration,
1280 currently hardcoded.
1283 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1286 meson: build gir even when cross-compiling if introspection was enabled explicitly
1287 This can be made to work in certain circumstances when
1288 cross-compiling, so default to not building g-i stuff
1289 when cross-compiling, but allow it if introspection was
1290 enabled explicitly via -Dintrospection=enabled.
1291 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1293 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1295 * gst/rtsp-server/rtsp-session.c:
1296 rtsp-session: clean up comment extra-timeout
1298 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1300 * gst/rtsp-server/rtsp-client.c:
1301 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1302 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1303 from the RTSP context.
1306 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1308 * gst/rtsp-server/rtsp-client.c:
1309 * gst/rtsp-server/rtsp-media.c:
1310 * gst/rtsp-server/rtsp-media.h:
1311 rtsp-client: Lock shared media
1312 For shared media we got race conditions. Concurrently rtsp clients might
1313 suspend or unsuspend the shared media and thus change the state without
1314 the clients expecting that.
1315 By introducing a lock that can be taken by callers such as rtsp_client
1316 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1317 to handle the media sequentially thus allowing one client to finish its
1318 rtsp call before another client calls on the same media.
1319 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1322 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1324 * gst/rtsp-server/rtsp-session.c:
1325 rtsp-session: add property extra-timeout
1326 Extra time to add to the timeout, in seconds. This only
1327 affects the time until a session is considered timed out
1328 and is not signalled in the RTSP request responses.
1329 Only the value of the timeout property is signalled in the
1332 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1334 * gst/rtsp-server/rtsp-stream.c:
1335 rtsp-stream : fix race condition in send_tcp_message
1336 If one thread is inside the send_tcp_message function and are done
1337 sending rtp or rtcp messages so the n_outstanding variable is zero
1338 however have not exit the loop sending the messages. While sending its
1339 messages, transports have been added or removed to the transport list,
1340 so the cache should be updated. If now an additional thread comes to
1341 the function send_tcp_message and trying to send rtp messages it will
1342 first destroy the rtp cache that is still being iterated trough by the
1346 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1355 * examples/.gitignore:
1356 * examples/Makefile.am:
1358 * gst/rtsp-server/.gitignore:
1359 * gst/rtsp-server/Makefile.am:
1360 * gst/rtsp-sink/Makefile.am:
1361 * pkgconfig/.gitignore:
1362 * pkgconfig/Makefile.am:
1364 * tests/Makefile.am:
1365 * tests/check/Makefile.am:
1366 Remove autotools build
1368 Maybe we can now use the meson pkgconfig module
1369 for .pc files? (Does it support uninstalled now?)
1371 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1373 * tests/check/gst/client.c:
1374 client: fix test mem leak in attach_rate_tweaking_probe
1376 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1378 * tests/check/gst/media.c:
1379 media: remove memleak in test test_media_seek
1381 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1383 * tests/check/gst/rtspserver.c:
1384 rtspserver: Remove memleak in test test_double_play
1386 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1388 * gst/rtsp-server/rtsp-media.c:
1389 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1391 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1393 * gst/rtsp-server/rtsp-media.c:
1394 * tests/check/gst/rtspserver.c:
1395 rtsp-media: Unblock all streams
1396 When unsuspending and going to PLAYING, unblock all streams instead of
1397 only those that are linked (the linked streams are the ones for which
1398 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1399 pushing buffers on unlinked streams.
1400 This change is because playback using single-threaded demuxers like
1401 matroska-demux could be blocked if SETUP was not called for all media.
1402 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1403 gstflvdemux, qtdemux, and matroska-demux) will handle
1404 GST_FLOW_NOT_LINKED automatically.
1407 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1409 * gst/rtsp-server/rtsp-media.c:
1410 * tests/check/gst/rtspserver.c:
1411 rtsp-media: Wait on async when needed.
1412 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1413 In the unit test the pause from adjust_play_mode will cause a preroll
1414 and after that async-done will be produced.
1415 Without this patch there are no one consuming this async-done and when
1416 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1417 wait for async-done. But then it wrongly find the async-done prodused by
1418 adjus_play_mode and continue executing without waiting for the preroll
1421 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1423 * gst/rtsp-server/rtsp-client.c:
1424 rtsp-client: RTP Info when completed_sender
1425 Change condition that should be fulfilled regarding RTPInfo.
1426 Replace !gst_rtsp_media_is_receive_only with
1427 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1428 for a sender pipeline that is complete. Only then a RTPInfo should
1430 gst_rtsp_media_is_receive_only gives different answears depending on
1432 If Describe is called wth URL+options for backchannel SDP will give only
1433 audio and only backchannel a=sendonly
1434 If Describe is called on URL+options that gives both audio and video
1435 direction from server to client, pipelines are created. Thus
1436 receive_only will return false, even though Setup only would setup
1438 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1439 streams are complete.
1441 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1443 * gst/rtsp-server/rtsp-client.c:
1444 * tests/check/gst/client.c:
1445 rtsp-client: RTP Info exists conditionally in PLAY
1446 If RTP Info is missing and it is not a receiver only, eg. audio
1447 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1448 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1449 Since 1.14 there is audio backchannel support. Thus RTP-info is
1450 conditional now. When audio backchannel only mode, there is no RTP-info.
1453 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1455 * examples/test-onvif-client.c:
1456 test-onvif-client: remove unused query
1458 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1460 * gst/rtsp-server/rtsp-client.c:
1461 rtsp-client: RTP Info must exist in PLAY response
1462 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1465 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1467 * examples/test-onvif-client.c:
1468 test-onvif-client: perform accurate seeks
1469 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1470 Also, modify how we compute the position: position queries in
1471 PAUSED mode fail to account for the newly-prerolled frame, leading
1472 to frame skips when performing seeks in that state. Instead,
1473 compute the current position from the last sample.
1475 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1477 * gst/rtsp-server/rtsp-client.c:
1478 * gst/rtsp-server/rtsp-media.c:
1479 * gst/rtsp-server/rtsp-media.h:
1480 * tests/check/gst/rtspserver.c:
1481 Use complete streams for scale and speed.
1482 Without this patch it's always stream0 that is used to get segment event
1483 that is used to set scale and speed. This even if client not doing SETUP
1484 for stream0. At least in suspend mode reset this not working since then
1485 it's just random if send_rtp_sink have got any segment event. There are
1486 no check if send_rtp_sink for stream0 got any data before media is
1487 prerolled after PLAY request.
1489 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1491 * examples/test-onvif-server.c:
1492 * examples/test-onvif-server.h:
1493 examples/onvif-server: fix werror build with clang
1494 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:346:65: warning: implicit conversion from enumeration type 'const GstSegmentFlags' to different enumeration type 'GstSeekFlags' [-Wenum-conversion]
1495 self->incoming_segment->format, self->incoming_segment->flags,
1496 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1497 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1498 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1500 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1501 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1503 <scratch space>:77:1: note: expanded from here
1506 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1507 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1509 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1510 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1512 <scratch space>:9:1: note: expanded from here
1515 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1516 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1517 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1519 <scratch space>:12:1: note: expanded from here
1523 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1526 meson: Don't generate doc cache when no plugins are enabled
1527 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1529 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1531 * examples/test-onvif-client.c:
1532 test-onvif-client: stdin is not defined in MSVC
1534 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1536 * gst/rtsp-server/rtsp-media.c:
1537 rtsp-media: add missing Since tag
1539 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1541 * examples/test-onvif-client.c:
1542 test-onvif-client: STDIN_FILENO is not portable
1543 If not defined, define it to _fileno(stdin) on Windows, 0
1546 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1548 * examples/test-onvif-server.c:
1549 test-onvif-server: downgrade logging
1551 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1553 * examples/meson.build:
1554 * examples/test-onvif-client.c:
1555 * examples/test-onvif-server.c:
1556 examples: add ONVIF client / server example
1558 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1560 * gst/rtsp-server/rtsp-client.c:
1561 * gst/rtsp-server/rtsp-media.c:
1562 rtsp-client: define all seek accuracy flags from setup_play_mode
1563 We then pass those to adjust_play_mode, which needs to operate
1564 on the "final" seek flags, as previously the code in rtsp-media
1565 was assuming that accuracy seek flags (accurate / key_unit) should
1566 not be set if the flags passed to the seek method were already set.
1568 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1570 * gst/rtsp-server/rtsp-media-factory-uri.c:
1571 * gst/rtsp-server/rtsp-media.c:
1572 rtsp-media: Try to get dynamic payloaders by name from their bin first
1573 First try "pay", then "pay_%s" (where %s == pad name). And only then
1574 fall back to the code that simply takes the first payloader that is
1576 The current code usually works (but is racy) because it will always take
1577 the payloader that was last added (due to g_list_prepend() when adding
1578 elements) in pad-added and that's usually the correct one. But if a new
1579 payloader is added between pad-added and us trying to get it, we would
1580 get the wrong payloader.
1582 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1584 * tests/check/gst/client.c:
1585 client test: expect any port in transport
1586 setup_multicast_client sets a 5000-5010 range for the client
1587 ports, it is incorrect to expect the transport to always use
1591 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1593 * tests/check/gst/onvif.c:
1594 onvif tests: use g_cond_wait() correctly
1595 g_cond_wait() has to be called in a loop until required conditions
1599 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1601 * gst/rtsp-server/rtsp-stream.c:
1602 rtsp-stream: Not wait on receiver streams when pre-rolling
1603 Without this patch there are problem pre-rolling when using audio back
1605 Without this patch a probe will be created for all streams including
1606 the stream for audio backchannel. To pre-roll all this pads have to
1607 receive data. Since the stream for audio backchannel is a receiver this
1609 The solution is to never create any probes for streams that are for
1610 incomming data and instead set them as blocking already from beginning.
1612 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1614 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1615 * gst/rtsp-server/rtsp-onvif-media.c:
1616 onvif-media: fix "void function returning a value" compiler warning
1618 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1620 * gst/rtsp-server/rtsp-media.c:
1621 rtsp-media: make sure streams are blocked when sending seek
1622 The recent ONVIF work exposed a race condition when dealing with
1623 multiple streams: one of the sinks may preroll before other streams
1624 have started flushing. This led to the pipeline posting async-done
1625 prematurely, when some streams were actually still in the middle
1626 of performing a flushing seek. The newly-added code looks up a
1627 sticky segment event on the first stream in order to respond to
1628 the PLAY request with accurate Scale and Speed headers. In the
1629 failure condition, the first stream was flushing, and thus had
1630 no sticky segment event, leading to the PLAY request failing,
1631 and in turn the test.
1633 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
1636 * gst/rtsp-server/rtsp-media-factory-uri.h:
1639 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1641 * gst/rtsp-server/rtsp-client.c:
1642 * gst/rtsp-server/rtsp-client.h:
1643 * gst/rtsp-server/rtsp-media.c:
1644 * gst/rtsp-server/rtsp-media.h:
1645 * gst/rtsp-server/rtsp-onvif-client.c:
1646 * gst/rtsp-server/rtsp-onvif-client.h:
1647 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1648 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1649 * gst/rtsp-server/rtsp-onvif-media.c:
1650 * gst/rtsp-server/rtsp-onvif-server.h:
1651 * gst/rtsp-server/rtsp-stream.c:
1652 * gst/rtsp-server/rtsp-stream.h:
1653 * tests/check/gst/media.c:
1654 * tests/check/gst/onvif.c:
1655 * tests/check/meson.build:
1656 onvif: Implement and test the Streaming Specification
1657 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
1659 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1661 * gst/rtsp-server/rtsp-client.c:
1662 * gst/rtsp-server/rtsp-client.h:
1663 rtsp-client: add gst_rtsp_client_get_stream_transport()
1664 This will be used in the onvif tests in order to validate the
1665 data transmitted over TCP: for streaming to continue after a
1666 data message has been provided to client->send_func, the client
1667 is responsible for marking the message as sent on the relevant
1670 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1672 * gst/rtsp-server/rtsp-client.c:
1673 client: Scale implies TRICK_MODE
1675 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1677 * gst/rtsp-server/rtsp-client.c:
1678 client: compare booleans, not pointers to them
1680 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
1682 * gst/rtsp-server/rtsp-media.c:
1683 * gst/rtsp-server/rtsp-stream.c:
1684 * tests/check/gst/media.c:
1685 Reverse playback support
1686 GStreamer plays segment from stop to start when doing reverse playback.
1687 RTSP implies that media should be played from start of Range header to
1688 its stop. Hence we swap start and stop times before passing them to
1690 Also make gst_rtsp_stream_query_stop always return value that can be
1691 used as stop time of Range header.
1693 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
1695 * gst/rtsp-server/rtsp-client.c:
1696 * gst/rtsp-server/rtsp-media.c:
1697 * gst/rtsp-server/rtsp-media.h:
1698 * tests/check/gst/client.c:
1699 rtsp-client: add support for Scale and Speed header
1700 Add support for the RTSP Scale and Speed headers by setting the rate in
1701 the seek to (scale*speed). We then check the resulting segment for rate
1702 and applied rate, and use them as values for the Speed and Scale headers
1704 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1706 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
1708 * gst/rtsp-server/rtsp-client.c:
1709 * gst/rtsp-server/rtsp-client.h:
1710 rtsp-client: allow sub classes to adjust the seek
1711 Adds a new virtual function, adjust_play_mode(), that allows
1712 sub classes to adjust the seek done on the media. The sub class can
1713 modify the values of the the seek flags and the rate.
1714 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1716 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
1718 * gst/rtsp-server/rtsp-media.c:
1719 * gst/rtsp-server/rtsp-media.h:
1720 * gst/rtsp-server/rtsp-stream.c:
1721 * gst/rtsp-server/rtsp-stream.h:
1722 * tests/check/gst/media.c:
1723 rtsp-media: allow specifying rate when seeking
1724 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
1725 caller to specify the rate for the seek. Also added functions in
1726 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
1727 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1729 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
1733 meson: Bump minimal GLib version to 2.44
1734 This means we can use some newer features and get rid of some
1735 boilerplate code using the G_DECLARE_* macros.
1736 As discussed on IRC, 2.44 is old enough by now to start depending on it.
1738 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1740 * docs/libs/.gitignore:
1741 * docs/libs/Makefile.am:
1742 * docs/libs/gst-rtsp-server-docs.sgml:
1743 * docs/libs/gst-rtsp-server-sections.txt:
1744 * docs/libs/gst-rtsp-server.types:
1745 docs: remove obsolete gtk-doc related files
1747 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1749 * gst/rtsp-sink/gstrtspclientsink.c:
1750 doc: remove xml from comments
1752 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
1754 * docs/gst_plugins_cache.json:
1756 docs: Stop building the doc cache by default
1757 And update the cache
1758 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
1760 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
1762 * docs/gst_plugins_cache.json:
1763 docs: Update plugins documentation cache
1765 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
1768 * gst/rtsp-server/rtsp-context.c:
1769 * gst/rtsp-server/rtsp-session-pool.c:
1770 doc: Fix some docstrings
1772 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
1778 * docs/gst_plugins_cache.json:
1781 * docs/plugin-index.md:
1782 * docs/plugin-sitemap.txt:
1785 * docs/version.entities.in:
1786 * gst/rtsp-server/meson.build:
1787 * gst/rtsp-sink/meson.build:
1789 * meson_options.txt:
1790 docs: Port to hotdoc
1792 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1794 * gst/rtsp-server/rtsp-auth.c:
1795 * gst/rtsp-server/rtsp-client.h:
1796 rtsp-server: Fix various Since markers
1798 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1800 * gst/rtsp-server/rtsp-media.c:
1801 * gst/rtsp-server/rtsp-sdp.c:
1802 * gst/rtsp-server/rtsp-session-media.c:
1803 * gst/rtsp-server/rtsp-stream.c:
1804 rtsp-server: Add various Since: 1.14 markers
1806 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1808 * gst/rtsp-server/rtsp-media-factory.c:
1809 * gst/rtsp-server/rtsp-media.c:
1810 * gst/rtsp-server/rtsp-stream-transport.c:
1811 * gst/rtsp-server/rtsp-stream.c:
1812 rtsp-server: Add various missing Since: 1.16 markers
1814 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
1816 * gst/rtsp-sink/gstrtspclientsink.c:
1817 rtspclientsink: Set async-handling=false for the internal bins
1818 Without this we can easily run into a race condition with async state changes:
1819 - the pipeline is doing an async state change
1820 - we set the internal bins to PLAYING but that's ignored because an
1821 async state change is currently pending
1822 - the async state change finishes but does not change the state of the
1823 internal bins because of locked_state==TRUE
1824 - the internal bins stay in PAUSED forever
1826 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1828 * gst/rtsp-sink/gstrtspclientsink.c:
1829 rtspclientsink: Use write_messages() API to send buffer lists in one go
1830 And to write messages with multiple memories also via writev().
1832 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
1834 * gst/rtsp-server/rtsp-client.c:
1835 * gst/rtsp-server/rtsp-client.h:
1836 * gst/rtsp-server/rtsp-server-object.h:
1837 * gst/rtsp-server/rtsp-server.c:
1838 rtsp-client: Handle Content-Length limitation
1839 Add functionality to limit the Content-Length.
1840 API addition, Enhancement.
1841 Define an appropriate request size limit and reject requests
1842 exceeding the limit with response status 413 Request Entity Too Large
1845 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1852 === release 1.16.0 ===
1854 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1860 * gst-rtsp-server.doap:
1864 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
1866 * gst/rtsp-sink/gstrtspclientsink.c:
1867 rtspclientsink: Notify the stream transport about each written message
1868 Otherwise it will never try to send us the next one: it tries to keep
1869 exactly one message in-flight all the time.
1870 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
1871 in the client sink we always write data out synchronously.
1873 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
1875 * gst/rtsp-server/rtsp-stream.c:
1876 rtsp_server: Free thread pool before clean transport cache
1877 If not waiting for free thread pool before clean transport caches, there
1878 can be a crash if a thread is executing in transport list loop in
1879 function send_tcp_message.
1880 Also add a check if priv->send_pool in on_message_sent to avoid that a
1881 new thread is pushed during wait of free thread pool. This is possible
1882 since when waiting for free thread pool mutex have to be unlocked.
1884 === release 1.15.90 ===
1886 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
1892 * gst-rtsp-server.doap:
1896 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
1898 * gst/rtsp-server/rtsp-stream.c:
1899 rtsp-stream: Add support for GCM (RFC 7714)
1902 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
1904 * gst/rtsp-server/rtsp-session-pool.c:
1905 session pool: fix missing klass-> in klass->create_session
1907 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1910 g-i: pass --quiet to g-ir-scanner
1911 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
1912 that we get even if everything works just fine.
1913 We still get g-ir-scanner warnings and compiler warnings if
1914 we pass this option.
1916 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1919 g-i: silence 'nested extern' compiler warnings when building scanner binary
1920 We need a nested extern in our init section for the scanner binary
1921 so we can call gst_init to make sure GStreamer types are initialised
1922 (they are not all lazy init via get_type functions, but some are in
1923 exported variables). There doesn't seem to be any other mechanism to
1924 achieve this, so just remove that warning, it's not important at all.
1926 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1929 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
1931 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
1933 * gst/rtsp-server/rtsp-media.c:
1934 * tests/check/gst/media.c:
1935 rtsp-media: Handle set state when preparing.
1936 Handle the situation when a call to gst_rtsp_media_set_state is done
1937 when media status is preparing.
1938 Also add unit test for this scenario.
1939 The unit test simulate on a media level when two clients share a (live)
1941 Both clients have done SETUP and got responses. Now client 1 is doing
1942 play and client 2 is just closing the connection.
1943 Then without patch there are a problem when
1944 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
1945 And client2 is doing closing connection we can end up in a call
1946 to gst_rtsp_media_set_state when
1947 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
1948 shut down media is jumped over .
1949 With this patch and this scenario we wait until
1950 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
1951 execute after that and now we will execute the logic for
1954 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
1962 === release 1.15.2 ===
1964 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
1970 * gst-rtsp-server.doap:
1974 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
1976 * gst/rtsp-server/rtsp-media.c:
1977 * tests/check/gst/client.c:
1978 rtsp-media: Fix multicast use case with common media
1987 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
1989 * gst/rtsp-server/rtsp-client.c:
1990 * gst/rtsp-server/rtsp-stream.c:
1991 * gst/rtsp-server/rtsp-stream.h:
1992 rtsp-server: remove recursive behavior
1993 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
1995 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
1997 * gst/rtsp-server/rtsp-client.c:
1998 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
1999 And route all messages through the send_func if no send_messages_func
2001 We otherwise break backwards compatibility.
2003 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2005 * docs/libs/gst-rtsp-server-sections.txt:
2006 * gst/rtsp-server/rtsp-client.c:
2007 * gst/rtsp-server/rtsp-client.h:
2008 * gst/rtsp-server/rtsp-stream.c:
2009 rtsp-client: Add support for sending buffer lists directly
2010 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2012 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2014 * docs/libs/gst-rtsp-server-sections.txt:
2015 * gst/rtsp-server/rtsp-client.c:
2016 * gst/rtsp-server/rtsp-media.c:
2017 * gst/rtsp-server/rtsp-stream-transport.c:
2018 * gst/rtsp-server/rtsp-stream-transport.h:
2019 * gst/rtsp-server/rtsp-stream.c:
2020 * gst/rtsp-sink/gstrtspclientsink.c:
2021 rtsp-server: Add support for buffer lists
2022 This adds new functions for passing buffer lists through the different
2023 layers without breaking API/ABI, and enables the appsink to actually
2024 provide buffer lists.
2025 This should already reduce CPU usage and potentially context switches a
2026 bit by passing a whole buffer list from the appsink instead of
2027 individual buffers. As a next step it would be necessary to
2028 a) Add support for a vector of data for the GstRTSPMessage body
2029 b) Add support for sending multiple messages at once to the
2030 GstRTSPWatch and let it be handled internally
2031 c) Adding API to GOutputStream that works like writev()
2032 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2034 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2036 * gst/rtsp-server/rtsp-client.c:
2037 client: Fix crash in close handler
2038 The close handler could trigger a crash because it invalidated the
2039 watch_context while still leaving a source attached to it which would be
2040 cleaned up at a later point.
2042 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2044 * gst/rtsp-server/rtsp-stream.c:
2045 rtsp-stream: Use cached address when allocating sockets
2046 If an address/port was previously decided upon (ex: multicast in the
2047 SDP), then use that instead of re-creating another one
2048 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2050 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2052 * gst/rtsp-server/rtsp-media.c:
2053 rtsp-media: Fix race codition in finish_unprepare
2054 The previous fix for race condition around finish_unprepare where the
2055 function could be called twice assumed that the status wouldn't change
2056 during execution of the function. This assumption is incorrect as the
2057 state may change, for example if an error message arrives from the
2059 Instead a flag keeping track on whether the finish_unprepare function
2060 is currently executing is introduced and checked.
2061 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2063 === release 1.15.1 ===
2065 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2071 * gst-rtsp-server.doap:
2075 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2077 * gst/rtsp-server/rtsp-stream.c:
2078 Add source elements to the pipeline before activation
2079 In plug_src we changed the element state before adding it to
2080 the owner container. This prevented the pipeline from intercepting
2081 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2082 to assign a custom task pool.
2083 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2085 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2088 Automatic update of common submodule
2089 From ed78bee to 59cb678
2091 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2093 * examples/test-appsrc.c:
2094 examples: test-appsrc: fix coding style error
2096 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2098 * examples/test-appsrc.c:
2099 examples: test-appsrc: fix buffer leak
2101 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2103 * gst/rtsp-server/rtsp-media.c:
2104 rtsp-media: Update priv->blocked when linked streams are unblocked.
2105 Media is considered to be blocked when all streams that belong to
2106 that media are blocked.
2107 This patch solves the problem of inconsistent updates of
2108 priv->blocked that are not synchronized with the media state.
2110 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2112 * gst/rtsp-server/rtsp-media.c:
2113 rtsp-media: Don't block streams before seeking
2114 Before the seek operation is performed on media, it's required that
2115 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2116 At this stage, all transport parts (transport sinks) have been successfully
2117 added to the pipeline and there is no need for blocking the streams.
2119 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2121 * tests/check/gst/rtspserver.c:
2122 tests: rtspserver: Add shared media test case for TCP
2124 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2126 * gst/rtsp-server/rtsp-stream.c:
2127 rtsp-stream: Use seqnum-offset for rtpinfo
2128 The sequence number in the rtpinfo is supposed to be the first RTP
2129 sequence number. The "seqnum" property on a payloader is supposed to be
2130 the number from the last processed RTP packet. The sequence number for
2131 payloaders that inherit gstrtpbasepayload will not be correct in case of
2132 buffer lists. In order to fix the seqnum property on the payloaders
2133 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2134 "seqnum-offset" from the "stats" property contains the value of the
2135 very first RTP packet in a stream. The server will, however, try to look
2136 at the last simple in the sink element and only use properties on the
2137 payloader in case there no sink elements yet, and by looking at the last
2138 sample of the sink gives the server full control of which RTP packet it
2139 looks at. If the payloader does not have the "stats" property, "seqnum"
2140 is still used since "seqnum-offset" is only present in as part of
2141 "stats" and this is still an issue not solved with this patch.
2142 Needed for gst-plugins-base!17
2144 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2146 * gst/rtsp-server/rtsp-stream.c:
2147 rtsp-stream: Plug memory leak
2148 Attaching a GSource to a context will increase the refcount. The idle
2149 source will never be free'd since the initial reference is never
2152 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2155 Add Gitlab CI configuration
2156 This commit adds a .gitlab-ci.yml file, which uses a feature
2157 to fetch the config from a centralized repository. The intent is
2158 to have all the gstreamer modules use the same configuration.
2159 The configuration is currently hosted at the gst-ci repository
2160 under the gitlab/ci_template.yml path.
2161 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2163 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2166 * gst-rtsp-server.doap:
2167 Update git locations to gitlab
2169 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2171 * gst/rtsp-server/meson.build:
2172 meson: add new onvif types
2174 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2176 * gst/rtsp-server/meson.build:
2177 Add ONVIF subclass headers to the installed headers in meson.build too
2179 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2181 * gst/rtsp-server/rtsp-server-object.h:
2182 * gst/rtsp-server/rtsp-server.h:
2183 rtsp-server: Declare GstRTSPServer struct before anything else
2184 It's needed by all kinds of other headers, including the ones that are
2185 required for defining the GstRTSPServer struct itself and its API.
2187 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2189 * gst/rtsp-server/rtsp-onvif-client.h:
2190 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2191 * gst/rtsp-server/rtsp-onvif-media.h:
2192 * gst/rtsp-server/rtsp-onvif-server.h:
2193 Mark all ONVIF-specific subclasses as Since 1.14
2195 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2197 * gst/rtsp-server/Makefile.am:
2198 * gst/rtsp-server/meson.build:
2199 * gst/rtsp-server/rtsp-context.h:
2200 * gst/rtsp-server/rtsp-onvif-server.c:
2201 * gst/rtsp-server/rtsp-onvif-server.h:
2202 * gst/rtsp-server/rtsp-server-object.h:
2203 * gst/rtsp-server/rtsp-server-prelude.h:
2204 * gst/rtsp-server/rtsp-server.c:
2205 * gst/rtsp-server/rtsp-server.h:
2206 * gst/rtsp-server/rtsp-session.h:
2207 Include ONVIF types from single-include rtsp-server.h
2208 ... by actually making it a single-include header and moving everything
2209 related to the GstRTSPServer type to rtsp-server-object.h instead.
2210 Otherwise there are too many circular includes.
2211 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2213 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2215 * gst/rtsp-server/rtsp-client.c:
2216 * gst/rtsp-server/rtsp-latency-bin.c:
2217 * gst/rtsp-server/rtsp-stream.c:
2218 * gst/rtsp-server/rtsp-stream.h:
2219 rtsp-stream: use idle source in on_message_sent
2220 When the underlying layers are running on_message_sent, this sometimes
2221 causes the underlying layer to send more data, which will cause the
2222 underlying layer to run callback on_message_sent again. This can go on
2224 To break this chain, we introduce an idle source that takes care of
2225 sending data if there are more to send when running callback
2226 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2228 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2230 * gst/rtsp-server/rtsp-client.c:
2231 rtsp-client: Remove timeout GSource on cleanup
2232 Avoids ending up with races where a timeout would still be around
2233 *after* a client was gone. This could happen rather easily in
2234 RTSP-over-HTTP mode on a local connection, where each RTSP message
2235 would be sent as a different HTTP connection with the same tunnelid.
2236 If not properly removed, that timeout would then try to free again
2237 a client (and its contents).
2239 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2241 * gst/rtsp-server/Makefile.am:
2242 autotools: fix distcheck
2244 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2246 * gst/rtsp-server/Makefile.am:
2247 * gst/rtsp-server/meson.build:
2248 * gst/rtsp-server/rtsp-latency-bin.c:
2249 * gst/rtsp-server/rtsp-latency-bin.h:
2250 * gst/rtsp-server/rtsp-onvif-media.c:
2251 onvif: encapsulate onvif part into a bin
2252 ...and thus do not let onvif affect pipelines latency
2253 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2255 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2257 * tests/check/gst/client.c:
2258 tests: client: Avoid bind() failures in tests
2259 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2261 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2263 * gst/rtsp-server/rtsp-media-factory.c:
2264 * gst/rtsp-server/rtsp-media-factory.h:
2265 * gst/rtsp-server/rtsp-media.c:
2266 * gst/rtsp-server/rtsp-media.h:
2267 * gst/rtsp-server/rtsp-stream.c:
2268 * gst/rtsp-server/rtsp-stream.h:
2269 * tests/check/gst/client.c:
2270 * tests/check/gst/mediafactory.c:
2271 New property for socket binding to mcast addresses
2272 By default the multicast sockets are bound to INADDR_ANY,
2273 as it's not allowed to bind sockets to multicast addresses
2274 in Windows. This default behaviour can be changed by setting
2275 bind-mcast-address property on the media-factory object.
2276 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2278 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2281 * gst/rtsp-server/Makefile.am:
2282 * gst/rtsp-server/meson.build:
2283 * gst/rtsp-server/rtsp-address-pool.c:
2284 * gst/rtsp-server/rtsp-auth.c:
2285 * gst/rtsp-server/rtsp-client.c:
2286 * gst/rtsp-server/rtsp-context.c:
2287 * gst/rtsp-server/rtsp-media-factory-uri.c:
2288 * gst/rtsp-server/rtsp-media-factory.c:
2289 * gst/rtsp-server/rtsp-media.c:
2290 * gst/rtsp-server/rtsp-mount-points.c:
2291 * gst/rtsp-server/rtsp-params.c:
2292 * gst/rtsp-server/rtsp-permissions.c:
2293 * gst/rtsp-server/rtsp-sdp.c:
2294 * gst/rtsp-server/rtsp-server-prelude.h:
2295 * gst/rtsp-server/rtsp-server.c:
2296 * gst/rtsp-server/rtsp-session-media.c:
2297 * gst/rtsp-server/rtsp-session-pool.c:
2298 * gst/rtsp-server/rtsp-session.c:
2299 * gst/rtsp-server/rtsp-stream-transport.c:
2300 * gst/rtsp-server/rtsp-stream.c:
2301 * gst/rtsp-server/rtsp-thread-pool.c:
2302 * gst/rtsp-server/rtsp-token.c:
2304 libs: fix API export/import and 'inconsistent linkage' on MSVC
2305 Export rtsp-server library API in headers when we're building the
2306 library itself, otherwise import the API from the headers.
2307 This fixes linker warnings on Windows when building with MSVC.
2308 Fix up some missing config.h includes when building the lib which
2309 is needed to get the export api define from config.h
2310 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2312 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2314 * gst/rtsp-server/rtsp-media-factory.c:
2315 rtsp-media-factory: Add missing break statements
2316 This resulted in warnings/assertions whenever one accessed the
2317 max-mcast-ttl property.
2321 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2324 * meson_options.txt:
2325 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2327 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2330 * meson_options.txt:
2331 * tests/check/meson.build:
2332 meson: add option to disable build of rtspclientsink plugin
2334 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2336 * meson_options.txt:
2337 meson: re-arrange options
2339 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2342 * meson_options.txt:
2343 * tests/check/meson.build:
2344 * tests/meson.build:
2345 meson: Use feature option for tests option
2346 This was somehow missed the last time around.
2348 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2350 * gst/rtsp-server/meson.build:
2352 meson: Maintain macOS ABI through dylib versioning
2353 Requires Meson 0.48, but the feature will be ignored on older versions
2354 so it's safe to add it without bumping the requirement.
2356 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2358 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2360 * gst/rtsp-sink/meson.build:
2362 meson: add pkg-config file for the rtspclientsink plugin
2364 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2366 * gst/rtsp-server/rtsp-client.c:
2367 * tests/check/gst/client.c:
2368 rtsp-client: Avoid reuse of channel numbers for interleaved
2369 If a (strange) client would reuse interleaved channel numbers in
2370 multiple SETUP requests, we should not accept them. The channel
2371 numbers are used for looking up stream transports in the
2372 priv->transports hash table, and transports disappear from the table
2373 if channel numbers are reused.
2374 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2375 server to change the channel numbers suggested by the client.
2376 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2378 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2380 * tests/check/gst/client.c:
2381 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2382 Allow regex for matching transport header against expected pattern.
2383 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2385 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2387 * tests/check/meson.build:
2388 meson: There is no gstreamer-plugins-good-1.0.pc
2389 There is no installed version of that, only an uninstalled version.
2391 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2393 * gst/rtsp-server/rtsp-client.c:
2394 * tests/check/gst/stream.c:
2395 Fix indentation again
2397 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2399 * gst/rtsp-server/rtsp-client.c:
2400 * gst/rtsp-server/rtsp-stream.c:
2401 * gst/rtsp-server/rtsp-stream.h:
2402 * tests/check/gst/client.c:
2403 * tests/check/gst/stream.c:
2404 stream: Added a list of multicast client addresses
2405 When media is shared, the same media stream can be sent
2406 to multiple multicast groups. Currently, there is no API
2407 to retrieve multicast addresses from the stream.
2408 When calling gst_rtsp_stream_get_multicast_address() function,
2409 only the first multicast address is returned.
2410 With this patch, each multicast destination requested in SETUP
2411 will be stored in an internal list (call to
2412 gst_rtsp_stream_add_multicast_client_address()).
2413 The list of multicast groups requested by the clients can be
2414 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2415 There still exist some problems with the current implementation
2416 in the multicast case:
2417 1) The receiving part is currently only configured with
2418 regard to the first multicast client (see
2419 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2420 2) Secondly, of security reasons, some constraints should be
2421 put on the requested multicast destinations (see
2422 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2423 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2424 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2426 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2428 * gst/rtsp-server/rtsp-client.c:
2429 * gst/rtsp-server/rtsp-stream.c:
2430 * gst/rtsp-server/rtsp-stream.h:
2431 * tests/check/gst/client.c:
2432 stream: Choose the maximum ttl value provided by multicast clients
2433 The maximum ttl value provided so far by the multicast clients
2434 will be chosen and reported in the response to the current
2436 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2437 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2439 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2441 * gst/rtsp-server/rtsp-stream.c:
2442 * tests/check/gst/client.c:
2443 rtsp-stream: Don't require address pool in the transport specific case
2444 If "transport.client-settings" parameter is set to true, the client is
2445 allowed to specify destination, ports and ttl.
2446 There is no need for pre-configured address pool.
2447 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2448 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2450 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2452 * gst/rtsp-server/rtsp-client.c:
2453 * tests/check/gst/client.c:
2454 client: Don't reserve multicast address in the client setting case
2455 When two multicast clients request specific transport
2456 configurations, and "transport.client-settings" parameter is
2457 set to true, it's wrong to actually require that these two
2458 clients request the same multicast group.
2459 Removed test_client_multicast_invalid_transport_specific test
2460 cases as they wrongly require that the requested destination
2461 address is supposed to be present in the address pool, also in
2462 the case when "transport.client-settings" parameter is set to true.
2463 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2464 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2466 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2468 * gst/rtsp-server/rtsp-media-factory.c:
2469 * gst/rtsp-server/rtsp-media-factory.h:
2470 * gst/rtsp-server/rtsp-media.c:
2471 * gst/rtsp-server/rtsp-media.h:
2472 * gst/rtsp-server/rtsp-stream.c:
2473 * gst/rtsp-server/rtsp-stream.h:
2474 * tests/check/gst/mediafactory.c:
2475 Add new API for setting/getting maximum multicast ttl value
2476 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2477 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2479 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2481 * gst/rtsp-server/rtsp-stream.c:
2482 rtsp-stream: avoid duplicating the first multicast client
2483 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2484 clients were dynamically added and removed to the multicast
2485 udp sinks, as such we should no longer add a first client in
2486 set_multicast_socket_for_udpsink
2487 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2489 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2491 * gst/rtsp-server/rtsp-stream.c:
2492 Revert "rtsp-stream: avoid duplicating the first multicast client"
2493 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2494 Commits where accidentially squashed together
2496 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2498 * gst/rtsp-server/rtsp-client.c:
2499 * gst/rtsp-server/rtsp-media-factory.c:
2500 * gst/rtsp-server/rtsp-media-factory.h:
2501 * gst/rtsp-server/rtsp-media.c:
2502 * gst/rtsp-server/rtsp-media.h:
2503 * gst/rtsp-server/rtsp-stream.c:
2504 * gst/rtsp-server/rtsp-stream.h:
2505 * tests/check/gst/client.c:
2506 * tests/check/gst/mediafactory.c:
2507 Revert "Add new API for setting/getting maximum multicast ttl value"
2508 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2509 Commits where accidentially squashed together
2511 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2513 * gst/rtsp-server/rtsp-stream.c:
2514 * tests/check/gst/client.c:
2515 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2516 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2517 Commits where accidentially squashed together
2519 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2521 * gst/rtsp-server/rtsp-client.c:
2522 * gst/rtsp-server/rtsp-stream.c:
2523 * gst/rtsp-server/rtsp-stream.h:
2524 * tests/check/gst/client.c:
2525 * tests/check/gst/stream.c:
2526 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2527 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2528 Commits where accidentially squashed together
2530 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2532 * examples/test-auth-digest.c:
2533 examples: Fix indentation
2535 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2537 * gst/rtsp-server/rtsp-client.c:
2538 * gst/rtsp-server/rtsp-stream.c:
2539 * gst/rtsp-server/rtsp-stream.h:
2540 * tests/check/gst/client.c:
2541 * tests/check/gst/stream.c:
2542 stream: Choose the maximum ttl value provided by multicast clients
2543 The maximum ttl value provided so far by the multicast clients
2544 will be chosen and reported in the response to the current
2546 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2548 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2550 * gst/rtsp-server/rtsp-stream.c:
2551 * tests/check/gst/client.c:
2552 rtsp-stream: Don't require address pool in the transport specific case
2553 If "transport.client-settings" parameter is set to true, the client is
2554 allowed to specify destination, ports and ttl.
2555 There is no need for pre-configured address pool.
2556 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2558 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2560 * gst/rtsp-server/rtsp-client.c:
2561 * gst/rtsp-server/rtsp-media-factory.c:
2562 * gst/rtsp-server/rtsp-media-factory.h:
2563 * gst/rtsp-server/rtsp-media.c:
2564 * gst/rtsp-server/rtsp-media.h:
2565 * gst/rtsp-server/rtsp-stream.c:
2566 * gst/rtsp-server/rtsp-stream.h:
2567 * tests/check/gst/client.c:
2568 * tests/check/gst/mediafactory.c:
2569 Add new API for setting/getting maximum multicast ttl value
2570 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2572 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2574 * gst/rtsp-server/rtsp-stream.c:
2575 rtsp-stream: avoid duplicating the first multicast client
2576 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2577 clients were dynamically added and removed to the multicast
2578 udp sinks, as such we should no longer add a first client in
2579 set_multicast_socket_for_udpsink
2580 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2582 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2584 * gst/rtsp-server/Makefile.am:
2585 rtsp-server: Add gstreamer-base gir dir in autotools
2587 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2589 * gst/rtsp-server/rtsp-client.c:
2590 * gst/rtsp-server/rtsp-stream.c:
2591 rtsp-client: always allocate both IPV4 and IPV6 sockets
2592 multiudpsink does not support setting the socket* properties
2593 after it has started, which meant that rtsp-server could no
2594 longer serve on both IPV4 and IPV6 sockets since the patches
2595 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2597 When first connecting an IPV6 client then an IPV4 client,
2598 multiudpsink fell back to using the IPV6 socket.
2599 When first connecting an IPV4 client, then an IPV6 client,
2600 multiudpsink errored out, released the IPV4 socket, then
2601 crashed when trying to send a message on NULL nevertheless,
2602 that is however a separate issue.
2603 This could probably be fixed by handling the setting of
2604 sockets in multiudpsink after it has started, that will
2605 however be a much more significant effort.
2606 For now, this commit simply partially reverts the behaviour
2607 of rtsp-stream: it will continue to only create the udpsinks
2608 when needed, as was the case since the patches were merged,
2609 it will however when creating them, always allocate both
2610 sockets and set them on the sink before it starts, as was
2611 the case prior to the patches.
2612 Transport configuration will only error out if the allocation
2613 of UDP sockets fails for the actual client's family, this
2614 also downgrades the GST_ERRORs in alloc_ports_one_family
2615 to GST_WARNINGs, as failing to allocate is no longer
2617 https://bugzilla.gnome.org/show_bug.cgi?id=796875
2619 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2622 * meson_options.txt:
2623 meson: Convert common options to feature options
2624 These are necessary for gst-build to set options correctly. The
2625 remaining automagic option is cgroup support in examples.
2626 https://bugzilla.gnome.org/show_bug.cgi?id=795107
2628 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2630 * gst/rtsp-server/rtsp-stream.c:
2631 rtsp-stream: Slightly simplify locking
2633 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
2635 * gst/rtsp-server/rtsp-client.c:
2636 * gst/rtsp-server/rtsp-stream-transport.c:
2637 * gst/rtsp-server/rtsp-stream-transport.h:
2638 * gst/rtsp-server/rtsp-stream.c:
2639 Limit queued TCP data messages to one per stream
2640 Before, the watch backlog size in GstRTSPClient was changed
2641 dynamically between unlimited and a fixed size, trying to avoid both
2642 unlimited memory usage and deadlocks while waiting for place in the
2643 queue. (Some of the deadlocks were described in a long comment in
2645 In the previous commit, we changed to a fixed backlog size of 100.
2646 This is possible, because we now handle RTP/RTCP data messages differently
2647 from RTSP request/response messages.
2648 The data messages are messages tunneled over TCP. We allow at most one
2649 queued data message per stream in GstRTSPClient at a time, and
2650 successfully sent data messages are acked by sending a "message-sent"
2651 callback from the GstStreamTransport. Until that ack comes, the
2652 GstRTSPStream does not call pull_sample() on its appsink, and
2653 therefore the streaming thread in the pipeline will not be blocked
2654 inside GstRTSPClient, waiting for a place in the queue.
2655 pull_sample() is called when we have both an ack and a "new-sample"
2656 signal from the appsink. Then, we know there is a buffer to write.
2657 RTSP request/response messages are not acked in the same way as data
2658 messages. The rest of the 100 places in the queue are used for
2659 them. If the queue becomes full of request/response messages, we
2660 return an error and close the connection to the client.
2661 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2663 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
2665 * gst/rtsp-server/rtsp-client.c:
2666 rtsp-client: Use fixed backlog size
2667 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
2668 Preparation for the next commit, which changes to a different way of
2669 avoiding both deadlocks and unlimited memory usage with the watch
2672 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2674 * gst/rtsp-server/rtsp-media.c:
2675 rtsp-media: unref clock (if set) when finalizing
2676 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2678 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2680 * docs/libs/gst-rtsp-server-sections.txt:
2681 rtsp-media: add gst_rtsp_media_*_set_clock to docs
2682 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2684 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2686 * gst/rtsp-server/rtsp-media-factory.c:
2687 media-factory: unref old clock when setting new clock
2688 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2690 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
2692 * gst/rtsp-server/rtsp-media-factory.c:
2693 media-factory: unref clock in finalize
2694 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2696 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2698 * gst/rtsp-server/rtsp-onvif-media.c:
2699 rtsp-onvif-media: fix g-ir-scanner warnings
2701 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2704 .gitignore: add another example binary
2706 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
2708 * examples/meson.build:
2709 meson: add new test-appsrc2 example to meson build
2711 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2713 * examples/Makefile.am:
2714 examples: fix build of new test-appsrc2 example
2715 Need to link against libgstapp-1.0.
2717 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
2719 * examples/.gitignore:
2720 * examples/Makefile.am:
2721 * examples/test-appsrc2.c:
2722 examples: Add test-appsrc2
2723 Add an example of feeding both audio and video into an RTSP
2724 pipeline via appsrc.
2726 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
2728 * gst/rtsp-server/rtsp-client.c:
2729 client: Strip transport parts as whitespaces could be around commas
2730 https://bugzilla.gnome.org/show_bug.cgi?id=758428
2732 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
2734 * gst/rtsp-server/rtsp-stream.c:
2735 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
2736 Fix race when setting up source elements.
2737 Since we set the source element(s) to PLAYING state before hooking
2738 them up to the downstream funnel, it's possible for the source element
2739 to receive packets before we actually get to linking it to the funnel,
2740 in which case buffers would be pushed out on an unlinked pad, causing
2741 it to error out and stop receiving more data.
2742 We fix this by blocking the source's srcpad until we have linked it.
2743 https://bugzilla.gnome.org/show_bug.cgi?id=796160
2745 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
2747 * gst/rtsp-server/rtsp-stream.c:
2748 rtsp-stream: Fix mismatch between allowed and configured protocols
2749 https://bugzilla.gnome.org/show_bug.cgi?id=796679
2751 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
2753 * gst/rtsp-server/rtsp-stream.c:
2754 rtsp-stream: Emit a signal when the SRTP decoder is created
2755 https://bugzilla.gnome.org/show_bug.cgi?id=778080
2757 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
2759 * gst/rtsp-server/rtsp-stream.c:
2760 rtsp-stream: Don't require presence of sinks in _get_*_socket()
2761 Transport specific sink elements are added to the pipeline
2762 in PLAY request and sockets are already created in SETUP so
2763 it's actually wrong to require the presence of sinks in
2764 _get_*_socket() functions.
2765 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2767 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
2769 * gst/rtsp-server/rtsp-stream.c:
2770 rtsp-stream: Update transport for multicast clients as well
2771 If a multicast client requests different transport settings
2772 than the existing one make sure that this new transport
2773 configuruation is propagated to the multicast udp sink.
2774 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2776 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
2778 * gst/rtsp-server/rtsp-stream.c:
2779 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
2780 And not on unicast udp sinks
2781 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2783 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
2785 * gst/rtsp-server/rtsp-address-pool.c:
2786 * gst/rtsp-server/rtsp-auth.c:
2787 * gst/rtsp-server/rtsp-client.c:
2788 * gst/rtsp-server/rtsp-media-factory-uri.c:
2789 * gst/rtsp-server/rtsp-media-factory.c:
2790 * gst/rtsp-server/rtsp-media.c:
2791 * gst/rtsp-server/rtsp-mount-points.c:
2792 * gst/rtsp-server/rtsp-server.c:
2793 * gst/rtsp-server/rtsp-session-media.c:
2794 * gst/rtsp-server/rtsp-session-pool.c:
2795 * gst/rtsp-server/rtsp-session.c:
2796 * gst/rtsp-server/rtsp-stream-transport.c:
2797 * gst/rtsp-server/rtsp-stream.c:
2798 * gst/rtsp-server/rtsp-thread-pool.c:
2799 Update for g_type_class_add_private() deprecation in recent GLib
2801 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
2803 * gst/rtsp-server/rtsp-auth.c:
2804 * gst/rtsp-server/rtsp-media.c:
2805 * gst/rtsp-server/rtsp-sdp.c:
2806 * gst/rtsp-server/rtsp-stream.c:
2809 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
2811 * examples/Makefile.am:
2812 * examples/test-video-disconnect.c:
2813 examples: Add test-video-disconnect example
2814 Simple example which cuts off all clients 10 seconds
2815 after the first one connects.
2817 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2819 * docs/libs/gst-rtsp-server-sections.txt:
2820 * examples/test-auth-digest.c:
2821 * gst/rtsp-server/rtsp-auth.c:
2822 * gst/rtsp-server/rtsp-auth.h:
2823 rtsp-auth: Add support for parsing .htdigest files
2824 Passwords are usually not stored in clear text, but instead
2825 stored already hashed in a .htdigest file.
2826 Add support for parsing such files, add API to allow setting
2827 a custom realm in RTSPAuth, and update the digest example.
2828 https://bugzilla.gnome.org/show_bug.cgi?id=796637
2830 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
2832 * gst/rtsp-sink/gstrtspclientsink.c:
2833 * gst/rtsp-sink/gstrtspclientsink.h:
2834 rtspclientsink: fix waiting for multiple streams
2835 We were previously only ever waiting for a single stream to notify it's
2836 blocked status through GstRTSPStreamBlocking. Actually count streams to
2838 Fixes rtspclientsink sending SDP's without out some of the input
2840 https://bugzilla.gnome.org/show_bug.cgi?id=796624
2842 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2844 * docs/libs/gst-rtsp-server-sections.txt:
2845 docs: add missing auth methods
2847 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2849 * gst/rtsp-server/rtsp-stream.c:
2850 rtsp-stream: only create funnel if it didn't exist already.
2851 This precented using multiple protocols for the same stream.
2852 https://bugzilla.gnome.org/show_bug.cgi?id=796634
2854 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2856 * examples/meson.build:
2857 meson: build auth-digest example
2859 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
2861 * gst/rtsp-server/rtsp-client.c:
2862 * gst/rtsp-server/rtsp-media.c:
2863 * gst/rtsp-server/rtsp-sdp.c:
2864 * gst/rtsp-server/rtsp-session-media.c:
2865 * gst/rtsp-server/rtsp-stream-transport.c:
2866 Get payloader stats only for the sending streams
2867 Get/set payloader properties only for streams that actually
2868 contain a payloader element.
2869 https://bugzilla.gnome.org/show_bug.cgi?id=796523
2871 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
2873 * gst/rtsp-server/Makefile.am:
2874 Makefile: Don't hardcode libtool for g-i build
2875 Similar to the other commits in core/base/bad
2877 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
2879 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2880 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
2881 https://bugzilla.gnome.org/show_bug.cgi?id=796229
2883 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
2885 * gst/rtsp-sink/gstrtspclientsink.c:
2886 rtspclientsink: Don't deadlock in preroll on early close
2887 If the connection is closed very early, the flushing
2888 marker might not get set and rtspclientsink can get
2889 deadlocked waiting for preroll forever.
2890 https://bugzilla.gnome.org/show_bug.cgi?id=786961
2892 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2895 * meson_options.txt:
2896 meson: Update option names to omit disable_ and with- prefixes
2897 Also yield common options to the outer project (gst-build in our case)
2898 so that they don't have to be set manually.
2900 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2903 meson: use -Wl,-Bsymbolic-functions where supported
2904 Just like the autotools build.
2906 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2909 * tests/check/Makefile.am:
2910 configure: check for -good and -bad plugins only in uninstalled setup
2911 Avoids confusing configure messages looking or a -good .pc file
2913 Also use plugindir variables that common macros set while at it.
2914 https://bugzilla.gnome.org/show_bug.cgi?id=795466
2916 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
2918 * gst/rtsp-server/rtsp-client.c:
2919 rtsp-client: Fix session timeout
2920 When streaming data over TCP then is not the keep-alive
2921 functionality working.
2922 The reason is that the function do_send_data have changed
2923 to boolean but the code is still checking the received result
2924 from send_func with GST_RTSP_OK.
2925 The result is that a successful send_func will always lead to
2926 that do_send_data is returning false and the keep-alive will
2928 https://bugzilla.gnome.org/show_bug.cgi?id=795321
2930 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2932 * docs/libs/gst-rtsp-server-sections.txt:
2933 * gst/rtsp-server/rtsp-media.c:
2934 * gst/rtsp-server/rtsp-sdp.c:
2935 * gst/rtsp-server/rtsp-stream.c:
2936 * gst/rtsp-server/rtsp-stream.h:
2937 * gst/rtsp-sink/gstrtspclientsink.c:
2938 * gst/rtsp-sink/gstrtspclientsink.h:
2939 Implement support for ULP Forward Error Correction
2940 In this initial commit, interface is only exposed for RECORD,
2941 further work will be needed in rtspsrc to support this for
2943 https://bugzilla.gnome.org/show_bug.cgi?id=794911
2945 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2947 * gst/rtsp-server/rtsp-onvif-media.c:
2948 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
2949 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
2950 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
2951 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
2952 the opposite, just like the ONVIF standard.
2953 Let's follow those RFCs as we're doing RTSP here, and add a property at
2954 a later time if needed to switch to the SDP RFC behaviour.
2955 https://bugzilla.gnome.org/show_bug.cgi?id=793964
2957 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2960 Automatic update of common submodule
2961 From 3fa2c9e to ed78bee
2963 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
2965 * gst/rtsp-server/rtsp-client.c:
2966 * gst/rtsp-server/rtsp-media-factory.c:
2967 * gst/rtsp-server/rtsp-media.c:
2968 * gst/rtsp-server/rtsp-stream.c:
2969 * tests/check/gst/rtspclientsink.c:
2970 gst: Run everything through gst-indent again
2972 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
2974 * gst/rtsp-server/rtsp-media.c:
2975 * tests/check/gst/media.c:
2976 rtsp-media: query the position on active streams if media is complete
2977 If the media is complete, i.e. one or more streams have been configured
2978 with sinks, then we want to query the position on those streams only.
2979 A query on an incomplete stream may return a position that originates from
2981 https://bugzilla.gnome.org/show_bug.cgi?id=794964
2983 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2985 * gst/rtsp-sink/gstrtspclientsink.c:
2986 rtspclientsink: make sure not to use freed string
2987 Set transport string to NULL after freeing it, so that
2988 at worst we get a NULL pointer if constructing a new
2989 transport string fails (which shouldn't really fail here).
2990 Also check return value of that, just in case.
2993 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2995 * gst/rtsp-server/rtsp-client.c:
2996 rtsp-client: do not free string passed to take_header
2998 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3000 * gst/rtsp-server/rtsp-stream.c:
3001 rtsp-stream: do not take lock in request_aux_receiver
3002 Added it right before pushing the previous commit, it is
3003 incorrect and deadlocks because this function gets called
3004 from the join_bin thread, which already holds the lock,
3005 that's the reason why request_aux_sender didn't take the
3008 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3010 * docs/libs/gst-rtsp-server-sections.txt:
3011 * gst/rtsp-server/rtsp-media-factory.c:
3012 * gst/rtsp-server/rtsp-media-factory.h:
3013 * gst/rtsp-server/rtsp-media.c:
3014 * gst/rtsp-server/rtsp-media.h:
3015 * gst/rtsp-server/rtsp-stream.c:
3016 * gst/rtsp-server/rtsp-stream.h:
3017 rtsp-server: add API to enable retransmission requests
3018 "do-retransmission" was previously set when rtx-time != 0,
3019 which made no sense as do-retransmission is used to enable
3020 the sending of retransmission requests, where as rtx-time
3021 is used by the peer to enable storing of buffers in order
3022 to respond to retransmission requests.
3023 rtsp-media now also provides a callback for the
3024 request-aux-receiver signal.
3025 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3027 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3029 * gst/rtsp-sink/gstrtspclientsink.c:
3030 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3031 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3033 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3035 * gst/rtsp-sink/gstrtspclientsink.c:
3036 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3037 This in order to be able to decrypt the RTCP backchannel
3038 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3040 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3042 * gst/rtsp-server/rtsp-client.c:
3043 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3044 When sending back an encrypted RTCP back channel, it is useful
3045 for the client to know the encryption key.
3046 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3048 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3050 * gst/rtsp-server/rtsp-client.c:
3051 * gst/rtsp-server/rtsp-stream.c:
3052 * gst/rtsp-server/rtsp-stream.h:
3053 rtsp-stream: extract handle_keymgmt from rtsp-client
3054 rtspclientsink will also need to parse KeyMgmt headers
3055 sent by the server to decrypt the RTCP backchannel stream
3056 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3058 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3060 * gst/rtsp-sink/gstrtspclientsink.c:
3061 * tests/check/gst/rtspclientsink.c:
3062 rtspclientsink: Fix client ports for the RTCP backchannel
3063 This was broken since the work for delayed transport creation
3064 was merged: the creation of the transports string depends on
3065 calling stream_get_server_port, which only starts returning
3066 something meaningful after a call to stream_allocate_udp_sockets
3067 has been made, this function expects a transport that we parse
3068 from the transport string ...
3069 Significant refactoring is in order, but does not look entirely
3070 trivial, for now we put a band aid on and create a second transport
3071 string after the stream has been completed, to pass it in
3072 the request headers instead of the previous, incomplete one.
3073 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3075 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3077 * gst/rtsp-server/rtsp-client.c:
3078 rtsp-client:Error handling when equal http session cookie
3079 There are some clients that are sending same session cookie on random
3081 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3083 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3085 * gst/rtsp-server/rtsp-media-factory-uri.c:
3086 rtsp-media-factory-uri: Fix compilation with latest GLib
3087 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3088 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3089 data->factory = g_object_ref (factory);
3092 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3100 === release 1.14.0 ===
3102 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3108 * gst-rtsp-server.doap:
3112 === release 1.13.91 ===
3114 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3120 * gst-rtsp-server.doap:
3124 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3126 * gst/rtsp-server/Makefile.am:
3127 * gst/rtsp-server/meson.build:
3128 * gst/rtsp-server/rtsp-address-pool.h:
3129 * gst/rtsp-server/rtsp-auth.h:
3130 * gst/rtsp-server/rtsp-client.h:
3131 * gst/rtsp-server/rtsp-context.h:
3132 * gst/rtsp-server/rtsp-media-factory-uri.h:
3133 * gst/rtsp-server/rtsp-media-factory.h:
3134 * gst/rtsp-server/rtsp-media.h:
3135 * gst/rtsp-server/rtsp-mount-points.h:
3136 * gst/rtsp-server/rtsp-onvif-client.h:
3137 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3138 * gst/rtsp-server/rtsp-onvif-media.h:
3139 * gst/rtsp-server/rtsp-onvif-server.h:
3140 * gst/rtsp-server/rtsp-params.h:
3141 * gst/rtsp-server/rtsp-permissions.h:
3142 * gst/rtsp-server/rtsp-sdp.h:
3143 * gst/rtsp-server/rtsp-server-prelude.h:
3144 * gst/rtsp-server/rtsp-server.h:
3145 * gst/rtsp-server/rtsp-session-media.h:
3146 * gst/rtsp-server/rtsp-session-pool.h:
3147 * gst/rtsp-server/rtsp-session.h:
3148 * gst/rtsp-server/rtsp-stream-transport.h:
3149 * gst/rtsp-server/rtsp-stream.h:
3150 * gst/rtsp-server/rtsp-thread-pool.h:
3151 * gst/rtsp-server/rtsp-token.h:
3152 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3153 We need different export decorators for the different libs.
3154 For now no actual change though, just rename before the release,
3155 and add prelude headers to define the new decorator to GST_EXPORT.
3157 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3159 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3160 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3161 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3163 === release 1.13.90 ===
3165 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3171 * gst-rtsp-server.doap:
3175 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3177 * gst/rtsp-server/rtsp-media-factory.c:
3178 * gst/rtsp-server/rtsp-permissions.c:
3179 permissions: add Since tags and example for new API
3181 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3183 * docs/libs/gst-rtsp-server-sections.txt:
3184 * gst/rtsp-server/rtsp-media-factory.c:
3185 * gst/rtsp-server/rtsp-media-factory.h:
3186 * gst/rtsp-server/rtsp-permissions.c:
3187 * gst/rtsp-server/rtsp-permissions.h:
3188 * tests/check/gst/permissions.c:
3189 permissions: more bindings-friendly API
3190 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3192 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3195 meson: enable more warnings
3197 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3199 * gst/rtsp-server/rtsp-client.c:
3200 rtsp-client: Place netaddress meta on packets received via TCP
3201 This allows us to later map signals from rtpbin/rtpsource back to the
3202 corresponding stream transport, and allows to do keep-alive based on
3203 RTCP packets in case of TCP media transport.
3204 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3206 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3208 * gst/rtsp-sink/gstrtspclientsink.c:
3209 rtspclientsink: if OPEN failed, unqueue next command
3210 As READY_TO_PAUSED can no longer return async, the RECORD
3211 command will be queued before the OPEN command fails
3212 (for example in case the server could not be connected),
3213 and record then waits for ever.
3214 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3216 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3218 * gst/rtsp-sink/gstrtspclientsink.c:
3219 rtspclientsink: fix retrieval of custom payloader caps
3220 If a bin is passed as the custom payloader, the caps of
3221 its factory will be empty, the correct way to obtain the caps
3222 is to query its sinkpad.
3224 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3226 * gst/rtsp-sink/gstrtspclientsink.c:
3227 rtspclientsink: fix extra unref of custom payloader
3229 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3231 * gst/rtsp-sink/gstrtspclientsink.c:
3232 rspclientsink: fix recent code indentation
3234 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3236 * gst/rtsp-sink/gstrtspclientsink.c:
3237 rtspclientsink: add missing get_type prototype
3239 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3241 * gst/rtsp-sink/gstrtspclientsink.c:
3242 rtspclientsink: allow setting payloader as pad property
3243 This was a FIXME item, and can be quite useful, also
3244 allowing to specify payloader properties from the command
3245 line, which is always nice.
3246 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3248 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3250 * gst/rtsp-server/rtsp-media.c:
3251 rtsp-media: Replace g_print() log line
3252 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3254 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3256 * gst/rtsp-server/rtsp-media.c:
3257 * tests/check/gst/rtspclientsink.c:
3258 rtsp-media: fix RECORD getting stuck
3259 The test_record case was working because async=false had
3260 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3261 but that was incorrect, as it should not be needed.
3262 Removing async=false made the test fail as expected, this is
3263 fixed by not trying to preroll when preparing the media for
3264 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3265 and our peer will not start sending media until it has received
3266 a response to that request, and sent and received a response
3267 to RECORD as well, thus obviously preventing preroll.
3268 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3270 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3272 * gst/rtsp-server/rtsp-auth.c:
3273 rtsp-auth: fix set_tls_authentication_mode annotation
3275 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3277 * gst/rtsp-server/rtsp-onvif-media.c:
3278 rtp-server: remove redefined variable
3279 res is a boolean variable which is defined in the function scope and
3280 redefined, with no reason, in the loop scope. This patch removes the
3282 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3284 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3286 * gst/rtsp-server/rtsp-media.c:
3287 * gst/rtsp-server/rtsp-stream.c:
3288 * gst/rtsp-server/rtsp-stream.h:
3289 stream: Add functions for checking if stream is receiver or sender
3290 ...and replace all checks for RECORD in GstRTSPMedia which are really
3291 for "sender-only". This way the code becomes more generic and introducing
3292 support for onvif-backchannel later on will require no changes in
3295 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3297 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3298 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3299 onvif: Make requires_backchannel() public
3300 ...in order to let subclasses building the onvif part of the pipeline
3301 check whether backchannel shall be included or not.
3303 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3305 * gst/rtsp-server/rtsp-onvif-media.c:
3306 rtsp-server: Switch around sendonly/recvonly attributes
3307 They are wrong in the ONVIF streaming spec. The backchannel should be
3308 recvonly and the normal media should be sendonly: direction is always
3309 from the point of view of the SDP offerer (the server) according to
3312 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3314 * docs/libs/gst-rtsp-server-docs.sgml:
3315 * docs/libs/gst-rtsp-server-sections.txt:
3316 * examples/.gitignore:
3317 * examples/Makefile.am:
3318 * examples/test-onvif-backchannel.c:
3319 * gst/rtsp-server/Makefile.am:
3320 * gst/rtsp-server/rtsp-media.h:
3321 * gst/rtsp-server/rtsp-onvif-client.c:
3322 * gst/rtsp-server/rtsp-onvif-client.h:
3323 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3324 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3325 * gst/rtsp-server/rtsp-onvif-media.c:
3326 * gst/rtsp-server/rtsp-onvif-media.h:
3327 * gst/rtsp-server/rtsp-onvif-server.c:
3328 * gst/rtsp-server/rtsp-onvif-server.h:
3329 * gst/rtsp-server/rtsp-sdp.c:
3330 * gst/rtsp-server/rtsp-sdp.h:
3331 rtsp: Add support for ONVIF backchannel
3332 This adds a new RTSP server, client, media-factory and media subclass
3333 for handling the specifics of the backchannel. Ideally this later can be
3334 extended with other ONVIF specific features.
3336 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3338 * gst/rtsp-server/rtsp-media.c:
3339 rtsp-media: Add support for sending+receiving medias
3340 We need to add an appsrc/appsink in that case because otherwise the
3341 media bin will be a sink and a source for rtpbin, causing a pipeline
3343 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3345 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3351 === release 1.13.1 ===
3353 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3357 * gst-rtsp-server.doap:
3361 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3363 * gst/rtsp-server/rtsp-session-pool.c:
3364 session-pool: remove nullable return annotation
3365 create_watch can only return NULL from the API guards, no
3368 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3370 * gst/rtsp-server/rtsp-media-factory.c:
3371 * gst/rtsp-server/rtsp-media.c:
3372 set_clock functions: Add nullable annotations
3374 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3376 * gst/rtsp-server/rtsp-auth.c:
3377 * gst/rtsp-server/rtsp-client.c:
3378 * gst/rtsp-server/rtsp-media-factory.c:
3379 * gst/rtsp-server/rtsp-media.c:
3380 * gst/rtsp-server/rtsp-mount-points.c:
3381 * gst/rtsp-server/rtsp-server.c:
3382 * gst/rtsp-server/rtsp-session-media.c:
3383 * gst/rtsp-server/rtsp-session-pool.c:
3384 * gst/rtsp-server/rtsp-session.c:
3385 * gst/rtsp-server/rtsp-stream-transport.c:
3386 * gst/rtsp-server/rtsp-stream.c:
3387 * gst/rtsp-server/rtsp-thread-pool.c:
3388 All around: add annotations and API guards
3390 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3392 * tests/test-cleanup.c:
3393 test-cleanup: bind any port
3394 The meson test suite runs tests in parallel, trying to bind
3395 a single port made the test fail.
3397 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3400 meson: make version numbers ints and fix int/string comparison
3401 WARNING: Trying to compare values of different types (str, int).
3402 The result of this is undefined and will become a hard error
3403 in a future Meson release.
3405 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3407 * gst/rtsp-server/rtsp-context.c:
3408 gst_rtsp_context_get_current: add (skip) annotation
3409 The return value type is defined with G_DEFINE_POINTER_TYPE,
3410 and gi emits the following warning:
3411 Invalid non-constant return of bare structure or union; register as
3412 boxed type or (skip)
3414 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3416 * gst/rtsp-server/rtsp-client.c:
3417 rtsp-client: add type annotations
3418 gi doesn't seem to be able to figure out the type of the
3419 signal parameters when defined with G_DEFINE_POINTER_TYPE
3421 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3424 autotools: use -fno-strict-aliasing where supported
3425 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3427 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3430 meson: use -fno-strict-aliasing where supported
3431 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3433 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3435 * gst/rtsp-server/rtsp-mount-points.c:
3436 mount-points: bail out of loop again when matching mount points
3437 Previous patch led to us iterating the entire sequence. Bail out
3438 of the loop again if we have a match but are moving away from it.
3439 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3441 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3443 * tests/check/gst/mountpoints.c:
3444 tests: mountpoints: add more checks for mount point path matching
3445 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3447 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3449 * gst/rtsp-server/rtsp-mount-points.c:
3450 mount-points: fix matching of paths where there's also an entry with a common prefix
3451 e.g. with the following mount points
3455 _match() would not match /raw/video and /raw/snapshot correctly.
3456 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3458 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3460 * docs/libs/gst-rtsp-server-sections.txt:
3461 * gst/rtsp-server/rtsp-permissions.c:
3462 * gst/rtsp-server/rtsp-permissions.h:
3463 * tests/check/gst/permissions.c:
3464 permissions: add some new API to make this usable from bindings
3465 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3467 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3469 * gst/rtsp-server/rtsp-token.c:
3470 rtsp-token: annotate constructors for bindings
3471 This maps _new_empty() to _new(), which also makes RTSPToken()
3472 work properly now. Since this API wasn't usable from bindings
3473 before, this should hopefully be fine.
3474 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3476 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3478 * docs/libs/gst-rtsp-server-sections.txt:
3479 * gst/rtsp-server/rtsp-token.c:
3480 * gst/rtsp-server/rtsp-token.h:
3481 * tests/check/gst/token.c:
3482 rtsp-token: add some API to set fields from bindings
3483 The existing functions are all vararg-based and as such
3484 not usable from bindings.
3485 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3487 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3489 * tests/check/gst/rtspclientsink.c:
3490 * tests/check/gst/rtspserver.c:
3491 * tests/check/gst/sessionpool.c:
3492 * tests/check/gst/stream.c:
3493 tests: fix indentation
3496 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3498 * tests/check/gst/rtspserver.c:
3499 tests: rtspserver: fix another ref leak
3500 Even if this didn't show up in valgrind.
3502 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3504 * tests/check/gst/rtspclientsink.c:
3505 tests: rtspclientsink: fix leak
3507 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3509 * tests/check/gst/rtspserver.c:
3510 test: rtspserver: plug memory leak in test_no_session_timeout
3511 In test_no_session_timeout, unref the rtsp session object when the
3513 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3515 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3517 * gst/rtsp-sink/gstrtspclientsink.c:
3518 rtpsclientsink: Initialize and clear newly added mutex and cond
3519 While it *did* work, glib would automatically create new mutex and cond
3520 ... which never got freed
3522 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3524 * gst/rtsp-server/rtsp-stream.c:
3525 rtsp-stream: Set multicast TTL on the multicast sockets
3526 And not if we do unicast UDP.
3527 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3529 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3531 * gst/rtsp-server/rtsp-stream.c:
3532 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3533 In the multicast case (as in test-multicast, not test-multicast2), the
3534 address could be allocated/reserved (and thus set) already without
3535 allocating the actual socket. We need to allocate the socket here still
3536 instead of just claiming that it was already allocated.
3537 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3539 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3541 * gst/rtsp-sink/gstrtspclientsink.c:
3542 * gst/rtsp-sink/gstrtspclientsink.h:
3543 rtspclientsink: Use the new rtsp-stream API
3544 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3546 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3548 * gst/rtsp-sink/gstrtspclientsink.c:
3549 * gst/rtsp-sink/gstrtspclientsink.h:
3550 rtspclientsink: Wait until OPEN has been scheduled
3551 Make sure that the sink thread has started opening connection
3552 to the server before continuing.
3553 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3555 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3558 Automatic update of common submodule
3559 From e8c7a71 to 3fa2c9e
3561 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3563 * gst/rtsp-server/rtsp-media.c:
3564 * gst/rtsp-server/rtsp-session-media.c:
3565 * gst/rtsp-server/rtsp-stream.c:
3566 rtsp-server: Minor doc fixes
3569 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3572 * tests/Makefile.am:
3573 tests: disable all tests when --disable-tests is used
3574 Move conditional subdir include into top level.
3575 Based on patch by: Joel Holdsworth
3576 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3578 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3581 * meson_options.txt:
3582 * tests/meson.build:
3583 meson: build more tests and add options to disable tests and examples
3585 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3587 * gst/rtsp-server/rtsp-session.c:
3588 Fix build when -Werror=deprecated-declarations is on
3589 As gst_rtsp_session_next_timeout is deprecated.
3591 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:760:3: error: ‘gst_rtsp_session_next_timeout’ is deprecated: Use 'gst_rtsp_session_next_timeout_usec' instead [-Werror=deprecated-declarations]
3592 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3594 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3595 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3596 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3599 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3602 Automatic update of common submodule
3603 From 3f4aa96 to e8c7a71
3605 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3607 * tests/check/gst/media.c:
3608 check/media: Add seekability test case: not all streams are active
3609 Media contains two streams but only one is complete and prepared
3611 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3613 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3615 * gst/rtsp-server/rtsp-stream.c:
3616 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3617 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3619 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3621 * gst/rtsp-server/rtsp-media.c:
3622 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
3623 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3625 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
3628 meson: remove vs_module_defs_dir variable which is no longer needed
3630 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
3632 * gst/rtsp-server/rtsp-session.h:
3635 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
3638 * gst/rtsp-server/meson.build:
3640 * win32/common/libgstrtspserver.def:
3641 win32: remove .def file with exports
3642 They're no longer needed, symbol exporting is now explicit
3643 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
3645 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3648 autotools: stop controlling symbol visibility with -export-symbols-regex
3649 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
3650 This should result in consistent behaviour for the autotools and
3653 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
3655 * gst/rtsp-server/rtsp-media.h:
3656 * gst/rtsp-server/rtsp-server.h:
3657 * gst/rtsp-server/rtsp-session.c:
3658 * gst/rtsp-server/rtsp-session.h:
3659 rtsp-server: add missing GST_EXPORT and export deprecated funcs
3661 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
3663 * tests/check/gst/media.c:
3664 check: Add seekability testing on medias
3665 Make sure that once GstRTSPMedia are prepared they returned
3666 the expected seekability results
3667 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3669 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
3671 * docs/libs/gst-rtsp-server-sections.txt:
3672 * gst/rtsp-server/rtsp-media.c:
3673 * gst/rtsp-server/rtsp-stream.c:
3674 * gst/rtsp-server/rtsp-stream.h:
3675 * win32/common/libgstrtspserver.def:
3676 rtsp-media: Enable seeking query before pipeline is complete
3677 SDP are now provided *before* the pipeline is fully complete. In order
3678 to know whether a media is seekable or not therefore requires asking
3679 the invididual streams.
3680 API: gst_rtsp_stream_seekable
3681 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3683 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
3685 * gst/rtsp-server/rtsp-media.c:
3686 rtsp-media: Fix handling in default_unsuspend()
3687 Handle the case when streams are not blocked and media
3688 is suspended from PAUSED.
3689 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
3690 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3692 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
3694 * tests/check/gst/media.c:
3695 check/media: Fix thread pool leak.
3696 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
3697 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3699 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
3701 * gst/rtsp-server/rtsp-media.c:
3702 rtsp-media: Removed fakesink elements
3703 There is not need of adding fakesink elements to the media
3704 pipeline in the dynamic-payloader case.
3705 The media pipeline itself is dynamically updated with
3706 the receiver and sender parts that are based on the client
3707 transport information known after SETUP has been received.
3708 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
3709 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3711 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
3713 * gst/rtsp-server/rtsp-media.c:
3714 rtsp-media: Corrected ASYNC_DONE handling
3715 Media is complete when all the transport based parts are
3716 added to the media pipeline. At this point ASYNC_DONE is
3717 posted by the media pipeline and media is ready to enter
3719 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
3720 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3722 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
3724 * tests/check/gst/media.c:
3725 check/media: Check that prepared media can provide a SDP
3726 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
3728 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
3730 * gst/rtsp-server/rtsp-client.c:
3731 rtsp-client: Don't leak addr
3734 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
3736 * gst/rtsp-server/rtsp-client.c:
3737 * gst/rtsp-server/rtsp-session-media.c:
3738 * gst/rtsp-server/rtsp-stream.c:
3741 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
3743 * gst/rtsp-server/rtsp-media.c:
3744 rtsp-media: Don't unblock with remaining dynamic payloaders
3745 If we still have some dynamic paylaoders which haven't posted
3746 no-more-pads yet, don't go to PREPARED if one of the streams
3748 The risk was that we would end up not exposing/using all specified
3750 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
3751 then it will take a bit more time to start. But only if those 3
3752 conditions are present.
3753 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3755 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
3757 * gst/rtsp-server/rtsp-media.c:
3760 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
3762 * gst/rtsp-server/rtsp-media.c:
3763 rtsp-media: Don't set float on a gint64 variable
3764 Just use 0. Fixes 'undefined' behaviour from clang
3766 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
3768 * gst/rtsp-server/rtsp-media.c:
3769 rtsp-media: Fix previous commit
3770 We only want to count dynamic payloaders
3772 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
3774 * gst/rtsp-server/rtsp-media.c:
3775 * tests/check/gst/media.c:
3776 rtsp-media: Handle multiple dynamic elements
3777 If we have more than one dynamic payloader in the pipeline, we need
3778 to wait until the *last* one emits 'no-more-pads' before switching
3780 Failure to do so would result in a race where some of the streams
3781 wouldn't properly be prepared
3782 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3784 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
3786 * win32/common/libgstrtspserver.def:
3787 win32: Fix exported symbols list
3789 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3791 * gst/rtsp-server/rtsp-stream.c:
3792 rtsp-stream: Only update the RTP udpsink if it actually exists
3793 For send-only streams it does not exist, but the RTCP udpsink might.
3795 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
3797 * win32/common/libgstrtspserver.def:
3798 win32: Update exports
3800 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
3802 * gst/rtsp-server/rtsp-media.c:
3803 * gst/rtsp-server/rtsp-stream.c:
3804 * gst/rtsp-server/rtsp-stream.h:
3805 rtsp-media: seek on media pipelines that are complete
3806 Make sure that a seek is performed on pipelines that
3807 contain at least one sink element.
3808 Change-Id: Icf398e10add3191d104b1289de612412da326819
3809 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3811 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
3813 * gst/rtsp-server/rtsp-client.c:
3814 * gst/rtsp-server/rtsp-media.c:
3815 * gst/rtsp-server/rtsp-media.h:
3816 * gst/rtsp-server/rtsp-stream.c:
3817 * gst/rtsp-server/rtsp-stream.h:
3818 * tests/check/gst/client.c:
3819 * tests/check/gst/media.c:
3820 * tests/check/gst/rtspserver.c:
3821 * tests/check/gst/stream.c:
3822 Dynamically reconfigure pipeline in PLAY based on transports
3823 The initial pipeline does not contain specific transport
3824 elements. The receiver and the sender parts are added
3826 If the media is shared, the streams are dynamically
3827 reconfigured after each PLAY.
3828 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3830 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
3832 * gst/rtsp-server/rtsp-stream.c:
3833 rtsp-stream: obtain stream position from pad
3834 If no sinks have been added yet, obtain the current and
3835 the stop position of the stream from the send_src pad.
3836 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
3837 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3839 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
3841 * gst/rtsp-server/rtsp-session-media.c:
3842 * gst/rtsp-server/rtsp-session-media.h:
3843 rtsp-session-media: add function to get a list of transports
3844 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
3845 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3847 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
3849 * gst/rtsp-server/rtsp-stream.c:
3850 * gst/rtsp-server/rtsp-stream.h:
3851 rtsp-stream: add functions to get rtp and rtcp multicast sockets
3852 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
3853 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3855 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
3857 * gst/rtsp-server/rtsp-stream.c:
3858 stream: set async=sync=false only for RTCP appsink
3859 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
3860 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3862 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
3864 * gst/rtsp-server/rtsp-media.c:
3865 rtsp-media: return minimum value in query position case
3866 The minimum position should be returned as we are interested
3867 in the whole interval.
3868 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
3869 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3871 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
3873 * gst/rtsp-server/rtsp-session.c:
3874 * tests/check/gst/rtspserver.c:
3875 rtsp-session: Handle the case when timeout=0
3876 According to the documentation, a timeout of value 0 means
3877 that the session never timeouts. This adds handling of that.
3878 If timeout=0 we just return with a -1 from
3879 gst_rtsp_session_next_timeout_usec ().
3880 https://bugzilla.gnome.org/show_bug.cgi?id=785058
3882 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3884 * gst/rtsp-sink/gstrtspclientsink.c:
3885 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
3886 https://bugzilla.gnome.org/show_bug.cgi?id=785024
3888 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3890 * docs/libs/gst-rtsp-server-sections.txt:
3891 * gst/rtsp-server/rtsp-media-factory.c:
3892 docs: add media factory transport mode accessors
3893 and fix the documentation for the return value of the getter
3895 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
3897 * gst/rtsp-server/rtsp-client.c:
3898 rtsp-client: unref 'pipelined_requests' in finalize
3899 The hash table priv->pipelined_requests is not unref:ed in the
3900 finalize funktion. Make sure it is.
3901 https://bugzilla.gnome.org/show_bug.cgi?id=788704
3903 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
3905 * gst/rtsp-server/rtsp-media.c:
3906 rtsp-media: Initialize scalar variable
3909 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
3911 * win32/common/libgstrtspserver.def:
3912 win32: Update export file
3914 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
3916 * gst/rtsp-server/rtsp-client.c:
3917 * gst/rtsp-server/rtsp-media.c:
3918 * gst/rtsp-server/rtsp-media.h:
3919 Start support for RTSP 2.0
3920 This adds basic support for new 2.0 features, though the protocol is
3921 subposdely backward incompatible, most semantics are the sames.
3924 * version negotiation
3925 * pipelined requests support
3926 * Media-Properties support
3927 * Accept-Ranges support
3929 * gst_rtsp_media_seekable
3930 The RTSP methods that have been removed when using 2.0 now return
3932 https://bugzilla.gnome.org/show_bug.cgi?id=781446
3934 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3936 * gst/rtsp-server/rtsp-stream.c:
3937 stream: Use stream duration as stream-stop if segment was not configured with a stop
3938 Allowing client to know stream duration when no seeking happened.
3939 https://bugzilla.gnome.org/show_bug.cgi?id=783435
3941 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
3943 * gst/rtsp-server/rtsp-media-factory.c:
3944 rtsp-media-factory: Don't cache any media if NULL was returned as key
3945 The docs already mentioned this, but we actually stored it in the hash
3946 table with key==NULL and leaked its reference forever.
3948 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
3950 * gst/rtsp-sink/gstrtspclientsink.c:
3951 * gst/rtsp-sink/gstrtspclientsink.h:
3952 rtspclientsink: Use a mutex for protecting against concurrent send/receives
3953 This is a simple port of:
3954 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
3955 * c438545dc9e2f14f657bc0ef261fff726449867b
3956 * cd17c71dcea5c9310d21f1347c7520983e5869ac
3957 in gst-plugins-good.
3959 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
3961 * gst/rtsp-server/rtsp-sdp.c:
3962 sdp: fix Memory leak in error case
3963 https://bugzilla.gnome.org/show_bug.cgi?id=787059
3965 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3967 * pkgconfig/meson.build:
3968 meson: don't install -uninstalled.pc file
3969 https://bugzilla.gnome.org/show_bug.cgi?id=786457
3971 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
3974 Automatic update of common submodule
3975 From 48a5d85 to 3f4aa96
3977 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3979 * gst/rtsp-server/rtsp-client.c:
3980 rtsp-client: Fix typo in debug message
3982 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3985 meson: hide symbols by default unless explicitly exported
3987 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
3989 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3990 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
3991 Fixes meson warning about undefined @srcdir@.
3993 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
3995 * tests/meson.build:
3996 meson: skip tests on windows for now
3997 As we do in the other modules. As libgstcheck is currently not
3998 built on windows. Fixes "Fallback variable 'gst_check_dep' in
3999 the subproject 'gstreamer' does not exist"" Meson error.
4001 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4003 * gst/rtsp-server/rtsp-stream.c:
4004 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4005 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4006 multiudpsink's last-sample always comes from the payloader. Which
4007 is wrong if auxiliary streams are multiplexed in the same stream.
4008 So check the buffer's ssrc against the caps'ssrc before to use its
4009 seqnum. If not the same ssrc just use the payloader as done prior
4010 the commit above or when there is no last-sample yet.
4011 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4013 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4016 meson: Allow using glib as a subproject
4018 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4021 meson: fix with-package-name option
4022 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4024 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4027 Distribute meson_options.txt
4029 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4032 And config.h.meson is no longer dist either
4034 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4038 meson: config.h.meson is no longer needed
4040 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4042 * tests/check/meson.build:
4043 * tests/meson.build:
4044 meson: Fix building tests and activate them again
4046 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4048 * tests/check/meson.build:
4049 meson: Do not use path separator in test names
4050 Avoiding warnings like:
4051 WARNING: Target "elements/audioamplify" has a path separator in its name.
4053 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4056 * meson_options.txt:
4057 meson: add options to set package name and origin
4058 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4060 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4062 * gst/rtsp-server/rtsp-address-pool.h:
4063 * gst/rtsp-server/rtsp-auth.h:
4064 * gst/rtsp-server/rtsp-client.h:
4065 * gst/rtsp-server/rtsp-context.h:
4066 * gst/rtsp-server/rtsp-media-factory-uri.h:
4067 * gst/rtsp-server/rtsp-media-factory.h:
4068 * gst/rtsp-server/rtsp-media.h:
4069 * gst/rtsp-server/rtsp-mount-points.h:
4070 * gst/rtsp-server/rtsp-params.h:
4071 * gst/rtsp-server/rtsp-permissions.h:
4072 * gst/rtsp-server/rtsp-sdp.h:
4073 * gst/rtsp-server/rtsp-server.h:
4074 * gst/rtsp-server/rtsp-session-media.h:
4075 * gst/rtsp-server/rtsp-session-pool.h:
4076 * gst/rtsp-server/rtsp-session.h:
4077 * gst/rtsp-server/rtsp-stream-transport.h:
4078 * gst/rtsp-server/rtsp-stream.h:
4079 * gst/rtsp-server/rtsp-thread-pool.h:
4080 * gst/rtsp-server/rtsp-token.h:
4081 Mark symbols explicitly for export with GST_EXPORT
4083 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4086 * gst/rtsp-sink/Makefile.am:
4087 Remove plugin specific static build option
4088 Static and dynamic plugins now have the same interface. The standard
4089 --enable-static/--enable-shared toggle are sufficient.
4091 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4097 === release 1.12.0 ===
4099 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4105 * gst-rtsp-server.doap:
4109 === release 1.11.91 ===
4111 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4117 * gst-rtsp-server.doap:
4121 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4124 Automatic update of common submodule
4125 From 60aeef6 to 48a5d85
4127 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4129 * gst/rtsp-server/rtsp-media-factory.c:
4130 * gst/rtsp-server/rtsp-media.c:
4131 * gst/rtsp-server/rtsp-session.c:
4132 * gst/rtsp-server/rtsp-stream.c:
4133 gi: Fix some annotations and docstrings
4135 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4137 * gst/rtsp-server/meson.build:
4139 * meson_options.txt:
4142 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4146 Automatic update of common submodule
4147 From 39ac2f5 to 60aeef6
4149 === release 1.11.90 ===
4151 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4157 * gst-rtsp-server.doap:
4161 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4163 * examples/test-launch.c:
4164 examples: make test-launch pipeline shared by default as well
4166 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4168 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4169 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4170 Just the build dir is not going to work for srcdir!=builddir.
4172 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4175 meson: Update version
4177 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4182 === release 1.11.2 ===
4184 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4190 * gst-rtsp-server.doap:
4193 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4196 meson: dist meson build files
4197 Ship meson build files in tarballs, so people who use tarballs
4198 in their builds can start playing with meson already.
4200 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4202 * examples/test-record.c:
4203 examples/test-record: Add extra line to initial printout
4204 Add an example line of how to deliver a stream to the
4207 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4209 * gst/rtsp-server/rtsp-client.c:
4210 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4211 If there is no Content-Length header, no body would be allocated and the
4212 '\0' would also not be appended to the body.
4214 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4216 * gst/rtsp-server/rtsp-client.c:
4217 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4218 While they logically have 0 bytes length, GstRTSPConnection is appending
4219 a '\0' to everything making the size be 1 instead.
4221 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4226 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4228 * gst/rtsp-server/rtsp-session.c:
4229 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4230 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4233 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4238 === release 1.11.1 ===
4240 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4246 * gst-rtsp-server.doap:
4247 * win32/common/libgstrtspserver.def:
4250 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4252 * gst/rtsp-server/rtsp-stream.c:
4253 rtsp-stream: corrected if-statement in _get_server_port()
4254 This bug was accidentally introduced while fixing a segfault
4255 in _get_server_port() function.
4256 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4258 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4260 * gst/rtsp-server/rtsp-stream.c:
4261 * tests/check/gst/stream.c:
4262 rtsp-stream: fixed segmenation fault in _get_server_port()
4263 Calling function gst_rtsp_stream_get_server_port() results in
4264 segmenation fault in the RTP/RTSP/TCP case.
4265 Port that the server will use to receive RTCP makes only
4266 sense in the UDP case, however the function should handle
4267 the TCP case in a nicer way.
4268 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4270 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4272 * gst/rtsp-server/rtsp-media-factory.c:
4273 dosc: Fix a little typo
4274 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4276 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4278 * pkgconfig/Makefile.am:
4279 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4280 * pkgconfig/meson.build:
4281 meson: generate pkg-config -uninstalled pc files
4282 Generating those files is useful for users building the GStreamer stack
4283 using meson and having to link it to another project which is still
4284 using the autotools.
4285 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4287 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4289 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4290 pkgconfig: fix -uninstalled pc file
4291 pcfiledir was never defined so the paths were wrong.
4292 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4294 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4296 * gst/rtsp-server/rtsp-stream.c:
4297 * tests/check/gst/rtspserver.c:
4298 rtsp-stream: Fixed TCP transport case
4299 Make sure that the appsink element is actually added to
4300 the bin before trying to link it with the elements in it.
4301 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4303 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4309 Remove generated .spec file
4310 Likely extremely bitrotten, and we should not ship this anyway.
4312 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4315 Automatic update of common submodule
4316 From f980fd9 to 39ac2f5
4318 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4320 * gst/rtsp-server/rtsp-media.c:
4321 media: Fix pt map caps
4322 Since decryption is handled within rtpbin, all outcoming stream
4323 caps will be application/x-rtp (i.e. regular rtp)
4324 Fixes RECORD with SRTP streams
4326 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4328 * gst/rtsp-server/rtsp-media-factory.c:
4329 media-factory: Create media objects with the proper transport mode
4330 The function called immediately afterwards (collect_streams()) will
4331 need it to work properly
4333 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4335 * gst/rtsp-server/rtsp-auth.c:
4336 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4338 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4340 * gst/rtsp-server/rtsp-media-factory.c:
4341 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4342 We're going to put a pipeline into a pipeline otherwise, which is not
4345 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4347 * gst/rtsp-server/rtsp-media.c:
4348 media: Fix race condition around finish_unprepare() if called multiple time
4349 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4351 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4353 * gst/rtsp-sink/gstrtspclientsink.c:
4354 rtspclientsink: Don't leave stale pointer after unref
4355 Fix a warning on shutdown - don't keep a pointer to an
4356 alread-unreffed object.
4358 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4361 common: use https protocol for common submodule
4362 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4364 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4366 * gst/rtsp-server/rtsp-stream.c:
4367 stream: block the output of rtpbin instead of the source pipeline
4368 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4369 detection of the srtp rollover counter to add to the SDP.
4370 Unfortunately, it was incomplete for live pipelines where the logic
4371 blocks the source bin before creating the SDP and thus would never have
4372 the necessary informaiton to create a correct SDP with srtp encryption.
4373 Move the pad blocks to rtpbin's output pads instead so that the
4374 necessary information can be created before we need the information for
4376 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4378 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4380 * gst/rtsp-server/rtsp-client.c:
4381 rtsp-client: add IDLE timeout, before session exists
4382 The RTSP server will not timeout an idle RTSP connection
4383 (note this is different from doing timeout on a RTSP
4385 At least for Apache this is a problem when running RTSP over
4386 HTTPS since it uses one of the threads (there is a rather
4387 limited number) that are available for handling requests.
4388 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4390 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4395 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4397 * gst/rtsp-server/rtsp-stream.c:
4398 rtsp-stream: Set close-socket FALSE on UDP src:es
4399 With this RTSP server can use the sockets independent on the udpsrc
4401 When the udp src is finalized it will unref socket and when g_socket
4402 is finalized the socket will be closed.
4403 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4405 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4407 * gst/rtsp-sink/gstrtspclientsink.c:
4408 rtspclientsink: Move to new helper function to parse authentication responses
4409 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4411 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4413 * examples/Makefile.am:
4414 * examples/test-auth-digest.c:
4415 * gst/rtsp-server/rtsp-auth.c:
4416 * gst/rtsp-server/rtsp-auth.h:
4417 * win32/common/libgstrtspserver.def:
4418 rtsp-auth: Add support for Digest authentication
4419 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4421 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4424 * gst/rtsp-server/meson.build:
4426 * tests/check/meson.build:
4428 * win32/common/libgstrtspserver.def:
4429 Enable building with MSVC
4430 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4432 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4435 meson: gstreamer gst_check_dep does not exist on windows
4437 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4439 * gst/rtsp-server/rtsp-client.c:
4440 client: update do_send_message to match type GstRTSPClientSendFunc
4441 This type mismatch fails building with MSVC
4442 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4444 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4446 * gst/rtsp-server/rtsp-sdp.c:
4447 rtsp-sdp: Fix indentation
4449 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4451 * gst/rtsp-server/rtsp-media.c:
4452 rtsp-media: Only signal "new-state" if the state has actually changed
4453 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4455 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4457 * gst/rtsp-server/rtsp-client.c:
4458 * gst/rtsp-server/rtsp-client.h:
4459 client: emit signal in the beginning of each rtsp request
4460 These signals let the application validate the requests, configure the
4461 media/stream in a certain way and also generate error status code in
4462 case of error or bad request.
4463 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4465 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4468 meson: update version
4470 === release 1.11.0 ===
4472 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4477 === release 1.10.0 ===
4479 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4485 * gst-rtsp-server.doap:
4488 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4490 * tests/check/gst/rtspserver.c:
4491 * tests/check/gst/stream.c:
4492 tests: try to avoid using the same ports in different tests
4493 Causes problems with client multicast tests otherwise if
4494 tests are run in parallel.
4495 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4497 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4499 * tests/check/gst/client.c:
4500 tests: client: use fail_unless_equals_foo() for better failure reporting
4502 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4504 * gst/rtsp-server/rtsp-client.c:
4505 rtsp-client: Session filter in unwatch session
4506 Call session filter with filter_session_media as paramer in
4507 client_unwatch_session if using drop_backlog = FALSE.
4508 In client_unwatch_session its allowed to grow the watchs backlog.
4509 If using drop_backlog = FALSE and the backlog is full it will cause
4510 a deadlock when setting session media state to NULL
4511 if the backlog is not allowed to grow.
4512 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4514 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4517 meson: add fallbacks for gst modules
4520 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4522 * gst/rtsp-server/rtsp-client.c:
4523 rtsp-client: Fix factory leaking in find_media() in error cases
4524 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4526 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4528 * gst/rtsp-server/rtsp-stream.c:
4529 stream: Fix randomly missing streams from SDP with dynamic elements
4530 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4531 "pad-added" signal. In that case priv->srcpad could already have its caps,
4532 and they'll be sent to priv->send_src[0] pad. That means that when it
4533 connects "notify::caps" signal, that pad could already have received its
4534 caps and the signal won't be emitted anymore.
4535 In that case priv->caps stay to NULL and when building the SDP that stream
4536 gets ignored. Leading to missing video or audio when playing in client side.
4537 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4539 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4542 meson: update version
4544 === release 1.9.90 ===
4546 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4552 * gst-rtsp-server.doap:
4555 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4557 * gst/rtsp-server/rtsp-media-factory.c:
4558 * gst/rtsp-server/rtsp-media.c:
4559 * gst/rtsp-server/rtsp-stream.c:
4560 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4561 To prevent any possibly confusion with IPs or anything else.
4562 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4564 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4566 * gst/rtsp-server/rtsp-media-factory.c:
4567 * gst/rtsp-server/rtsp-media.c:
4568 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4569 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4571 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4574 configure: Depend on gstreamer 1.9.2.1
4576 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4580 Automatic update of common submodule
4581 From b18d820 to f980fd9
4583 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4587 Automatic update of common submodule
4588 From 6f2d209 to b18d820
4590 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4592 * gst/rtsp-server/rtsp-stream.c:
4593 rtsp-stream: Remove unused _locked() variant of a function
4594 It was added during refactoring.
4596 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4598 * gst/rtsp-server/rtsp-stream.c:
4599 stream: cosmetic cleanup
4600 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4602 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4604 * gst/rtsp-server/rtsp-stream.c:
4605 stream: Compare IP addresses case insensitive in more places
4606 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4608 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4611 * gst/rtsp-server/rtsp-stream.c:
4612 stream: Fix leaked joined_bin
4613 There is no need to keep a strong ref on it, and _leave_bin() was
4614 setting it to NULL before calling g_clear_object() so it was leaked.
4615 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4617 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4619 * gst/rtsp-server/rtsp-stream.c:
4620 rtsp-stream: Compare IP address strings case insensitive
4621 Otherwise IPv6 addresses might fail this comparision.
4623 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
4625 * gst/rtsp-server/rtsp-stream.c:
4626 rtsp-stream: Bind multicast sockets to ANY as before
4627 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
4629 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
4631 * gst/rtsp-server/rtsp-session.c:
4632 rtsp-session: Fix segfault when doing keep-alive after removing the session
4633 If keep-alive happens after removing the session but before finalizing the
4634 stream transport, we would segfault.
4635 https://bugzilla.gnome.org/show_bug.cgi?id=750544
4637 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
4639 * gst/rtsp-server/rtsp-stream.c:
4640 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
4641 Adding them later will cause deadlocks due to
4642 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
4643 2) adding the multicast sink
4644 3) waiting for it to get data to preroll again
4645 3) never happens because the queues after the tee are full.
4647 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
4649 * gst/rtsp-server/rtsp-stream.c:
4650 rtsp-stream: Fix up various multicast related issues
4652 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
4654 * tests/check/gst/stream.c:
4655 tests: Fix compilation
4657 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4659 * gst/rtsp-server/rtsp-client.c:
4660 * gst/rtsp-server/rtsp-stream.c:
4661 * tests/check/gst/stream.c:
4662 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
4663 This is basically reverting changes introduced in commit f62a9a7,
4664 because it was introducing various regressions:
4665 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
4666 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
4667 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
4668 - If a mcast client connects, it creates a new socket in SETUP to try to respect
4669 the destination/port given by the client in the transport, and overrides the
4670 socket already set on the udpsink element. That means that if we already had a
4671 client connected, the source address on the udp packets it receives suddenly
4673 - If a 2nd mcast client connects, the destination/port in its transport is
4674 ignored but its transport wasn't updated.
4675 What this patch does:
4676 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
4677 - Always have a tee+queue when udp is enabled. This could be optimized
4678 again in a later patch, but is more complicated. If no unicast clients
4679 connects then those elements are useless, this could be also optimized
4681 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
4682 seperated from those for unicast clients. Since we already support only
4683 one mcast address, we also create only one set of elements.
4684 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4686 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4688 * gst/rtsp-server/rtsp-stream.c:
4689 stream: factor our plug_src function
4690 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4692 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4694 * gst/rtsp-server/rtsp-stream.c:
4695 stream: factor out plug_sink function
4696 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4698 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4700 * gst/rtsp-server/rtsp-stream.c:
4701 stream: small documentation clarification
4702 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4704 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4706 * gst/rtsp-server/rtsp-stream.c:
4707 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
4708 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4710 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4712 * gst/rtsp-server/rtsp-stream.c:
4713 stream: Keep a ref on joined bin
4714 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4716 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4718 * gst/rtsp-server/rtsp-stream.c:
4719 stream: code cleanup
4720 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4722 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4724 * gst/rtsp-server/rtsp-stream.c:
4725 stream: small fix in error code path
4726 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4728 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4730 * gst/rtsp-server/rtsp-stream.c:
4731 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
4732 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
4733 but keeps unit tests.
4734 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4736 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
4741 === release 1.9.2 ===
4743 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
4749 * gst-rtsp-server.doap:
4752 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
4755 * examples/meson.build:
4757 * gst/rtsp-server/meson.build:
4758 * gst/rtsp-sink/meson.build:
4760 * pkgconfig/meson.build:
4761 * tests/check/meson.build:
4762 * tests/meson.build:
4763 Add support for Meson as alternative/parallel build system
4764 https://github.com/mesonbuild/meson
4766 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
4769 * tests/check/Makefile.am:
4770 build: silence error about pthread for 'make check' in osx
4771 Fixes "clang: error: argument unused during compilation: '-pthread'"
4773 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
4775 * gst/rtsp-server/rtsp-client.c:
4776 rtsp-client: Fix leaking of media in error cases
4777 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
4778 and myself to make the media refcounting a bit easier to follow.
4779 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4781 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4783 * gst/rtsp-server/rtsp-client.c:
4784 rtsp-client: Fix leaking of session in error cases
4785 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4787 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
4790 Automatic update of common submodule
4791 From f363b32 to f49c55e
4793 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
4798 === release 1.9.1 ===
4800 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4806 * gst-rtsp-server.doap:
4809 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4812 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
4813 https://bugzilla.gnome.org/show_bug.cgi?id=767463
4815 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4818 Automatic update of common submodule
4819 From ac2f647 to f363b32
4821 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4823 * gst/rtsp-server/rtsp-sdp.c:
4824 * gst/rtsp-server/rtsp-sdp.h:
4825 * gst/rtsp-server/rtsp-stream.c:
4826 * gst/rtsp-server/rtsp-stream.h:
4827 sdp: add rollover counters for all sender SSRC
4828 We add different crypto sessions in MIKEY, one for each sender
4829 SSRC. Currently, all of them will have the same security policy, 0.
4830 The rollover counters are obtained from the srtpenc element using the
4832 https://bugzilla.gnome.org/show_bug.cgi?id=730539
4834 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
4836 * gst/rtsp-server/rtsp-media-factory.h:
4837 * gst/rtsp-server/rtsp-server.h:
4838 docs: fix some typos
4840 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
4842 * gst/rtsp-server/Makefile.am:
4843 g-i: pass compiler env to g-ir-scanner
4844 It's what introspection.mak does as well. Should
4845 fix spurious build failures on gnome-continuous
4846 (caused by g-ir-scanner getting compiler details
4847 via python which is broken in some environments
4848 so passing the compiler details bypasses that).
4850 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
4852 * gst/rtsp-server/rtsp-session.c:
4853 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
4854 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
4855 https://bugzilla.gnome.org/show_bug.cgi?id=766619
4857 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
4859 * gst/rtsp-sink/gstrtspclientsink.c:
4860 rtspclientsink: Check return value of sscanf
4861 And just make sure we always have 0/0 if we have an error
4864 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
4866 * gst/rtsp-server/rtsp-stream.c:
4867 * tests/check/gst/rtspserver.c:
4868 * tests/check/gst/stream.c:
4869 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
4870 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
4871 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
4872 - Create unit test for shared media.
4873 https://bugzilla.gnome.org/show_bug.cgi?id=764744
4875 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4877 * gst/rtsp-server/rtsp-stream.c:
4878 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
4879 For IPv6 addresses, binding to a multicast group does not work on Linux
4880 either. Always bind to ANY and then later join the multicast group.
4881 https://bugzilla.gnome.org/show_bug.cgi?id=764679
4883 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
4886 Automatic update of common submodule
4887 From 6f2d209 to ac2f647
4889 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
4891 * gst/rtsp-server/rtsp-thread-pool.c:
4892 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
4893 Clarified why it is necessary to add source information to
4894 GstRTSPThreadImpl. See the reported bug in GLib:
4895 https://bugzilla.gnome.org/show_bug.cgi?id=720186
4896 for more information.
4897 https://bugzilla.gnome.org/show_bug.cgi?id=761702
4899 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
4901 * examples/Makefile.am:
4902 examples: Clean up CFLAGS/LDADD even more
4903 The internal .la should come first and is part of LDADD, as is
4906 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
4908 * examples/Makefile.am:
4909 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
4911 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
4913 * gst/rtsp-server/Makefile.am:
4914 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
4916 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
4918 * gst/rtsp-server/rtsp-client.c:
4919 * gst/rtsp-server/rtsp-media-factory.c:
4920 * gst/rtsp-server/rtsp-media-factory.h:
4921 * gst/rtsp-server/rtsp-media.c:
4922 * gst/rtsp-server/rtsp-media.h:
4923 * gst/rtsp-server/rtsp-sdp.c:
4924 * gst/rtsp-server/rtsp-stream.c:
4925 * gst/rtsp-server/rtsp-stream.h:
4926 rtsp-server: Implement clock signalling according to RFC7273
4927 For NTP and PTP clocks we signal the actual clock that is used and signal
4928 the direct media clock offset.
4929 For all other clocks we at least signal that it's the local sender clock.
4930 This allows receivers to know which clock was used to generate the media and
4931 its RTP timestamps. Receivers can then implement network synchronization,
4932 either absolute or at least relative by getting the sender clock rate directly
4933 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
4935 https://bugzilla.gnome.org/show_bug.cgi?id=760005
4937 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
4939 * gst/rtsp-sink/gstrtspclientsink.c:
4940 rtspclientsink: Add support for setting the multicast interface
4941 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4943 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4945 * gst/rtsp-server/rtsp-media-factory.c:
4946 * gst/rtsp-server/rtsp-media-factory.h:
4947 * gst/rtsp-server/rtsp-media.c:
4948 * gst/rtsp-server/rtsp-media.h:
4949 * gst/rtsp-server/rtsp-stream.c:
4950 * gst/rtsp-server/rtsp-stream.h:
4951 rtsp-media: Add support for setting the multicast interface
4952 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4954 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
4956 * gst/rtsp-sink/gstrtspclientsink.c:
4957 rtspclientsink: use new gst_element_class_add_static_pad_template()
4958 https://bugzilla.gnome.org/show_bug.cgi?id=763196
4960 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4965 === release 1.8.0 ===
4967 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
4973 * gst-rtsp-server.doap:
4976 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
4978 * gst/rtsp-server/rtsp-stream.c:
4979 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
4980 This would get us NO_PREROLL in the bin again and break seeking.
4981 Thanks to Carlos Rafael Giani for helping to debug this!
4982 https://bugzilla.gnome.org/show_bug.cgi?id=740509
4984 === release 1.7.91 ===
4986 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4992 * gst-rtsp-server.doap:
4995 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
4997 * gst/rtsp-server/rtsp-stream.c:
4998 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
4999 Without this, RECORD pipelines are broken because
5000 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5001 added later. Previously it was there earlier and due to NO_PREROLL caused the
5002 pipeline to preroll immediately
5003 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5004 as the corresponding code previously was only for PLAY pipelines.
5005 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5007 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5009 * gst/rtsp-server/rtsp-stream.c:
5010 rtsp-stream: Fix typo in the docstring
5011 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5013 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5015 * gst/rtsp-server/rtsp-stream.c:
5016 rtsp-stream: Disable multicast loopback for all our sockets
5017 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5018 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5019 loopback setting on the socket... while udpsink does which unfortunately has
5020 no effect here on Windows but on Linux.
5021 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5023 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5025 * tests/check/gst/stream.c:
5026 stream tests: added new tests
5027 Test a case when the address pool only contains multicast addresses
5028 and the client is requesting unicast udp.
5029 Added tests for multicast ports allocation.
5030 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5032 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5034 * gst/rtsp-server/rtsp-stream.c:
5035 rtsp-stream: Only bind multicast sockets to ANY on Windows
5036 On Linux it is still needed to bind to the multicast address
5037 to filter out random other packets, while on Windows binding
5038 to multicast addresses just fails.
5040 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5042 * gst/rtsp-server/rtsp-stream.c:
5043 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5044 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5045 addresses, which is something that used to work before. For unicast addresses
5046 if the pool contains none, we just allocate them as if there is no pool at
5048 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5050 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5052 * gst/rtsp-server/rtsp-client.c:
5053 * gst/rtsp-server/rtsp-stream.c:
5054 rtsp-server: Fix indentation
5056 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5058 * gst/rtsp-server/rtsp-stream.c:
5059 rtsp-stream: Don't bind the sockets to multicast addresses
5060 This works on Linux but fails completely on Windows. You're supposed
5061 to bind to ANY and then join the multicast group.
5062 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5064 === release 1.7.90 ===
5066 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5072 * gst-rtsp-server.doap:
5075 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5078 Automatic update of common submodule
5079 From b64f03f to 6f2d209
5081 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5083 * gst/rtsp-sink/gstrtspclientsink.c:
5084 * tests/check/gst/rtspclientsink.c:
5085 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5086 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5088 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5090 * tests/check/gst/media.c:
5091 * tests/check/gst/rtspclientsink.c:
5092 * tests/check/gst/rtspserver.c:
5093 * tests/check/gst/stream.c:
5094 tests: unit test fixes
5095 Removed port allocation test from the media suite.
5096 The port allocation failure is now in the stream suite.
5098 Make sure that the media is suspended after the DESCRIBE request
5099 before reconfiguring the UDP sinks.
5101 In the RECORD case we have to set async property to false
5102 for the appsink element in the test in order to make sure
5103 that the media pipeline doesn't hang in start_preroll().
5104 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5106 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5108 * gst/rtsp-server/rtsp-client.c:
5109 * gst/rtsp-server/rtsp-stream.c:
5110 * gst/rtsp-server/rtsp-stream.h:
5111 rtsp-stream: postpone UDP socket allocation until SETUP
5112 Postpone the allocation of the UDP sockets until we know
5113 what transport has been chosen by the client.
5114 Both unicast and multicast UDP sources are created in one
5116 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5118 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5120 * gst/rtsp-server/rtsp-stream.c:
5121 rtsp-stream: postpone the creation of the UDP sources
5122 Code refactoring: allocate the UDP ports after the sender and
5123 the reciver parts have been created.
5124 We postpone the creation of the UDP sources until the UDP
5125 ports have been allocated.
5126 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5128 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5130 * gst/rtsp-server/rtsp-stream.c:
5131 rtsp-stream: added function for setting UDP sources to PLAYING state
5132 Code refactoring: Introduced a function for setting UDP sources
5134 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5136 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5138 * gst/rtsp-server/rtsp-stream.c:
5139 rtsp-stream: added function for creating and configuring UDP sources
5140 Code refactoring: create and configure UDP sources in a separate function.
5141 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5143 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5145 * gst/rtsp-server/rtsp-stream.c:
5146 rtsp-stream: added function for RTP/RTCP socket configuration
5147 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5148 in a separate function.
5149 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5151 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5153 * gst/rtsp-server/rtsp-stream.c:
5154 rtsp-stream: added function for creating and configuring UDP sinks
5155 Code refactoring: create and configure UDP sinks in a separate function.
5156 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5158 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5160 * gst/rtsp-server/rtsp-stream.c:
5161 rtsp-stream: added helper function for creating the sender/receiver parts
5162 Code refactoring: introduced helper function for creating
5163 the receiver and the sender parts of the streaming pipeline.
5164 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5166 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5171 === release 1.7.2 ===
5173 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5179 * gst-rtsp-server.doap:
5182 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5184 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5185 uninstalled.pc: add support for non libtool build systems
5186 Currently the .la path is provided which requires to use libtool as
5187 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5188 It is fine as long as the application is built using libtool.
5189 So currently it is not possible to compile a GStreamer application
5190 within gst-uninstalled with CMake or other build system different
5192 This patch allows to do the following in gst-uninstalled env:
5193 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5194 gstreamer-rtsp-server-1.0)
5195 Previously it required to prepend libtool --mode=link
5196 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5198 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5200 * gst/rtsp-sink/gstrtspclientsink.c:
5201 rtspclientsink: remove check for impossible condition
5202 Goto error label checks stream to see if it needs to be unreferenced before
5203 returning, but this goto jumps happens before the stream is ever set, so it
5204 will always be NULL in this error label.
5207 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5209 * gst/rtsp-sink/gstrtspclientsink.c:
5210 rtspclientsink: clean switch statements
5211 Coverity demands for fallthrough statements to be clearly commented,
5212 to distinguish from accidental fall throughs. And it also needs all
5213 cases to finish with a break, even if the break is never going to be
5214 executed like in the case of a continue jump.
5218 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5220 * tests/check/Makefile.am:
5221 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5222 To get the CK_DEFAULT_TIMEOUT defined for all tests
5223 Also removes a 120 seconds timeout that was set as default
5224 explicitly in this module
5225 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5227 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5231 Automatic update of common submodule
5232 From 86e4663 to b64f03f
5234 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5236 * gst/rtsp-server/rtsp-media.c:
5237 rtsp-media: fix state_lock not locked again when preroll fails
5238 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5240 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5243 configure: Move plugin specific flags below all the others
5244 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5245 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5247 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5249 * gst/rtsp-sink/gstrtspclientsink.c:
5250 rtspclientsink: Simplify slightly using new -base API
5251 Use the new Mikey and SDP API in the base plugins libs
5252 to simplify some code.
5253 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5255 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5260 * gst/rtsp-sink/Makefile.am:
5261 * gst/rtsp-sink/gstrtspclientsink.c:
5262 * gst/rtsp-sink/gstrtspclientsink.h:
5263 * gst/rtsp-sink/plugin.c:
5264 * tests/check/Makefile.am:
5265 * tests/check/gst/rtspclientsink.c:
5266 rtspsink: Add rtspclientsink element
5267 Add an rtspclientsink element that accepts streams for which
5268 there is a registered payloader and sends them to
5269 an RTSP server using RECORD.
5270 Sending is synchronised to the pipeline clock. Payload-types
5271 are automatically selected. The 'new-payloader' signal is fired
5272 for custom configuration of payloaders when they are created.
5273 Can now stream a movie like this:
5275 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5276 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5278 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5279 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5280 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5282 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5284 * gst/rtsp-server/rtsp-stream.c:
5285 * gst/rtsp-server/rtsp-stream.h:
5286 rtsp-stream: Add functions for using rtsp-stream from the client
5287 Add a boolean to indicate that the rtsp-stream is running on the
5288 'client' side of an RTSP connection, for sending streams via
5289 RECORD. In that case, the roles of the client/server ports
5290 in transport setup are swapped.
5291 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5293 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5295 * gst/rtsp-server/rtsp-sdp.c:
5296 * gst/rtsp-server/rtsp-sdp.h:
5297 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5298 A new function that adds info from a GstRTSPStream into an SDP message.
5299 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5301 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5303 * gst/rtsp-server/rtsp-media.c:
5304 rtsp-media: Fix mutex beeing unlocked while they should be locked
5305 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5307 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5309 * gst/rtsp-server/rtsp-media-factory.c:
5310 rtsp-media-factory: add missing break in "clock" property setter
5313 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5315 * gst/rtsp-server/rtsp-stream.c:
5316 rtsp-stream: fixed assert during update transport
5317 When RTSP server trying update transport during multicast, it throws an
5318 assert. The assert is thrown because it is trying to get the parent of
5319 an non-existing funnel element.
5320 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5322 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5324 * gst/rtsp-server/rtsp-permissions.h:
5325 * gst/rtsp-server/rtsp-thread-pool.h:
5326 * gst/rtsp-server/rtsp-token.h:
5327 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5328 gtk-doc can handle static inline functions just fine these days,
5329 there's no need for this stuff any more.
5331 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5333 * gst/rtsp-server/rtsp-media.c:
5334 * gst/rtsp-server/rtsp-sdp.c:
5335 sdp: replace duplicated codes to call new base sdp apis
5336 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5338 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5340 * examples/test-netclock.c:
5341 test-netclock: Use the new API to configure a clock directly
5343 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5345 * gst/rtsp-server/rtsp-media-factory.c:
5346 * gst/rtsp-server/rtsp-media-factory.h:
5347 * gst/rtsp-server/rtsp-media.c:
5348 * gst/rtsp-server/rtsp-media.h:
5349 rtsp-media: Add API to directly configure a clock on the media pipelines
5351 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5353 * gst/rtsp-server/rtsp-media.c:
5354 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5356 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5358 * gst/rtsp-server/rtsp-media-factory.c:
5359 rtsp-media-factory: Add FIXME for 2.0
5361 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5363 * gst/rtsp-server/rtsp-stream.c:
5364 rtsp-stream: Fix indentation
5366 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5368 * gst/rtsp-server/rtsp-media.c:
5369 rtsp-media: Do not prepare media after media times out
5370 Deferred calls to start_prepare() can be deferred past the point until
5371 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5372 prepared to wait. Previously there was no lock and no check for this
5373 situation. This meant that a media could be prepared and unprepared
5374 simultaneously by two different threads. Now a lock is in place and a
5375 suitable check is done.
5376 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5378 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5380 * gst/rtsp-server/rtsp-client.c:
5381 * gst/rtsp-server/rtsp-media-factory.c:
5382 * gst/rtsp-server/rtsp-media-factory.h:
5383 * gst/rtsp-server/rtsp-media.c:
5384 * gst/rtsp-server/rtsp-media.h:
5385 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5386 Without TEARDOWN it might be desireable to keep the media running and continue
5387 sending data to the client, even if the RTSP connection itself is
5389 Only do this for session medias that have only UDP transports. If there's at
5390 least on TCP transport, it will stop working and cause problems when the
5391 connection is disconnected.
5392 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5394 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5399 === release 1.7.1 ===
5401 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5407 * gst-rtsp-server.doap:
5410 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5413 configure: Make -Bsymbolic check work with clang.
5414 Update the -Bsymbolic check with the version glib has. This version
5416 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5418 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5420 * gst/rtsp-server/rtsp-session-pool.c:
5421 rtsp-session-pool: Avoid dollar sign ($) in session ids
5422 Live555 in VLC strips off dollar signs and then gets very confused,
5423 we don't loose too much entropy by just skipping it.
5425 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5427 * gst/rtsp-server/rtsp-address-pool.h:
5428 * gst/rtsp-server/rtsp-auth.h:
5429 * gst/rtsp-server/rtsp-client.h:
5430 * gst/rtsp-server/rtsp-media-factory-uri.h:
5431 * gst/rtsp-server/rtsp-media-factory.h:
5432 * gst/rtsp-server/rtsp-media.h:
5433 * gst/rtsp-server/rtsp-mount-points.h:
5434 * gst/rtsp-server/rtsp-permissions.h:
5435 * gst/rtsp-server/rtsp-server.h:
5436 * gst/rtsp-server/rtsp-session-media.h:
5437 * gst/rtsp-server/rtsp-session-pool.h:
5438 * gst/rtsp-server/rtsp-session.h:
5439 * gst/rtsp-server/rtsp-stream-transport.h:
5440 * gst/rtsp-server/rtsp-stream.h:
5441 * gst/rtsp-server/rtsp-thread-pool.h:
5442 * gst/rtsp-server/rtsp-token.h:
5443 rtsp-server: Add g_autoptr() support to all types
5444 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5446 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5448 * gst/rtsp-server/rtsp-stream.c:
5449 rtsp-stream: fixed valgrind error
5450 Fixed the valgrind error in unit test. The UDP source created during
5451 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5453 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5455 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5459 Automatic update of common submodule
5460 From b319909 to 86e4663
5462 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5464 * gst/rtsp-server/rtsp-client.c:
5465 rtsp-client: suspend media during setup request
5466 SETUP request from clients needs to suspend the media to clear the
5467 prerolled buffers. Otherwise it will not affect the prerolled buffer
5468 and the prerolled buffers will be incorrect (for example block-size
5469 from setup request will not affect the prerolled buffer unless the
5470 media is suspended).
5471 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5473 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5475 * gst/rtsp-server/rtsp-stream.c:
5476 rtsp-stream: create stream pipeline based on transport
5477 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5478 only UDP is set as the transport protocol, it will not add the extra tee
5479 or queue element to the pipeline. Both these elements will be added, if
5480 it supports both TCP and UDP protocols. This improves the pipeline
5481 performance when one protocol is present.
5482 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5484 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5486 * gst/rtsp-server/rtsp-stream.c:
5487 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5488 Adding them when not needed will start some logic inside rtpbin that might be
5489 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5490 would start up a rtpjitterbuffer and behave in weird ways.
5491 We still set up the UDP sources for RTP receiving for a sender media to be
5492 able to receive any packets sent by the client for NAT traversal. They will
5493 all go to a fakesink though.
5494 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5495 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5496 receive ASYNC_DONE after a seek.
5497 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5499 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5501 * gst/rtsp-server/rtsp-stream.c:
5502 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5503 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5504 Previously we were only setting this for sender sockets, which caused looped
5505 back packets to be received on Windows if a multicast transport was used.
5507 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5509 * examples/test-record-auth.c:
5510 * examples/test-record.c:
5511 examples: Actually use the provided port in the record examples
5513 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5515 * examples/test-record-auth.c:
5516 test-record-auth: Add the option to build in TLS support
5518 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5520 * examples/test-auth.c:
5521 test-auth: Use an 'anonymous' user for unauthenticated default
5522 There's a comment on one of the resources that 'user' and 'admin'
5523 shouldn't even be able to see it, but they can if the default
5524 token is 'admin2', since that gives them access anyway.
5526 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5528 * examples/.gitignore:
5529 * examples/Makefile.am:
5530 * examples/test-record-auth.c:
5531 Add test-record-auth example
5533 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5535 * gst/rtsp-server/rtsp-client.c:
5536 * tests/check/gst/client.c:
5537 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5539 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5541 * gst/rtsp-server/rtsp-server.c:
5542 rtsp-server: Change the logic so we don't pop a NULL context
5543 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5544 will sometimes fail. This call is made before any context is pushed
5545 resulting in an attempt to pop a NULL context.
5546 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5548 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5550 * tests/check/gst/rtspserver.c:
5551 rtspserver: Add udp-mcast transport SETUP test
5552 Refactor utility functions in the test file so they can handle
5553 more than UDP and TCP as lower transport.
5554 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5556 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5558 * gst/rtsp-server/rtsp-stream.c:
5559 rtsp-stream: Always unref return value of gst_object_get_parent()
5560 Fixes a leak of a GstBin in the udp-mcast case.
5561 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5563 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5566 Automatic update of common submodule
5567 From b99800a to b319909
5569 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5572 Use new GST_ENABLE_EXTRA_CHECKS #define
5573 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5575 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5578 Automatic update of common submodule
5579 From 6babecd to b99800a
5581 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5584 Update GLib dependency to 2.40.0
5586 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5588 * examples/test-mp4.c:
5589 * gst/rtsp-server/rtsp-stream.c:
5590 stream: listen to sender ssrc signals
5591 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5593 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5596 common: update for new suppression
5597 Makes check-valgrind pass with glib 2.46
5599 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5601 * gst/rtsp-server/rtsp-media.c:
5602 rtsp-media: Take reference to media that will be prepared
5603 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5604 Later on a g_idle_source_new() is created and a pointer to the media
5605 object is passed as user data. If the media is freed before the idle
5606 source is dispatched the media object pointer is invalid, but the idle
5607 source callback expects it to still be valid. To fix this a reference to
5608 the media object is taken when registering the source callback function
5609 and a corresponding release of the reference is done when the souce is
5611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5613 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5615 * examples/test-launch.c:
5616 * examples/test-mp4.c:
5617 * examples/test-ogg.c:
5618 * examples/test-record.c:
5619 * examples/test-uri.c:
5620 rtsp-server: Fix memory leaks when context parse fails
5621 When g_option_context_parse fails, context and error variables are not getting free'd
5622 which results in memory leaks. Free'ing the same.
5623 And replacing g_error_free with g_clear_error, which checks if the error being passed
5624 is not NULL and sets the variable to NULL on free'ing.
5625 https://bugzilla.gnome.org/show_bug.cgi?id=753863
5627 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5632 === release 1.6.0 ===
5634 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5640 * gst-rtsp-server.doap:
5643 === release 1.5.91 ===
5645 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
5651 * gst-rtsp-server.doap:
5654 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
5656 * docs/libs/gst-rtsp-server-sections.txt:
5657 * gst/rtsp-server/rtsp-stream.c:
5658 stream: fix docs for recently-added get/set_buffer_size API
5659 https://bugzilla.gnome.org/show_bug.cgi?id=749095
5661 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
5663 * gst/rtsp-server/rtsp-media.c:
5664 rtsp-media: Don't crash on encrypted RTX SDP
5665 In parse_keymgmt(), don't mutate the input string that's been passed
5666 as const, especially since we might need the original value again if
5667 the same key info applies to multiple streams (RTX, for example).
5668 https://bugzilla.gnome.org/show_bug.cgi?id=754753
5670 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
5672 * examples/test-mp4.c:
5673 test-mp4: Support filenames with spaces in them. Error out on too few arguments
5675 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
5677 * examples/test-record.c:
5678 test-record: Check parameter count and print out help
5679 If no launch pipeline was supplied, print out some help
5681 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
5683 * gst/rtsp-server/rtsp-media.c:
5684 * gst/rtsp-server/rtsp-stream.c:
5685 * gst/rtsp-server/rtsp-stream.h:
5686 rtsp-stream: Implement UDP buffer size setting.
5687 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
5689 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
5690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
5692 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
5694 * gst/rtsp-server/rtsp-media.h:
5695 rtsp-media: Fix small typo causing gtk-doc to complain
5697 === release 1.5.90 ===
5699 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5705 * gst-rtsp-server.doap:
5708 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5710 * gst/rtsp-server/rtsp-media-factory.c:
5711 media-factory: get port number through gst_rtsp_url_get_port
5712 https://bugzilla.gnome.org/show_bug.cgi?id=753473
5714 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
5716 * tests/check/gst/media.c:
5717 media-test: Removing unnecessary assertion
5718 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5720 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5722 * gst/rtsp-server/rtsp-server.c:
5723 Document that source keeps a ref on server until it's destroyed
5724 https://bugzilla.gnome.org/show_bug.cgi?id=749227
5726 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5728 * tests/check/gst/media.c:
5729 media-test: Test for multiple dynamic payload
5730 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5732 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5734 * gst/rtsp-server/rtsp-media.c:
5735 media: Only add fakesink once per pipeline
5736 The intention is to prevent going PLAYING state before pads are created.
5737 If there was mutilple dynamic payload, it would leak few fakesink and
5738 actually prevent from ever reaching playing state.
5739 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5741 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5743 * gst/rtsp-server/rtsp-media.c:
5744 Revert "rtsp-media: Only add 1 fakesink per pipeline"
5745 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
5747 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5749 * gst/rtsp-server/rtsp-media.c:
5750 rtsp-media: Only add 1 fakesink per pipeline
5751 There should be only one fakesink per pipeline, not per dynpay. This
5752 would lead to element naming clash.
5754 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
5756 * gst/rtsp-server/rtsp-media.c:
5757 rtsp-media: assertion error due to wrong condition check
5758 In media to caps function, reserved_keys array is being used for variable i,
5759 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
5760 changed it to variable j
5761 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5763 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
5765 * gst/rtsp-server/rtsp-media.c:
5766 rtsp-media: Strip keys from the fmtp that we use internally in our caps
5767 Skip keys from the fmtp, which we already use ourselves for the
5768 caps. Some software is adding random things like clock-rate into
5769 the fmtp, and we would otherwise here set a string-typed clock-rate
5770 in the caps... and thus fail to create valid RTP caps
5771 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5773 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5775 * gst/rtsp-server/rtsp-thread-pool.c:
5776 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
5777 https://bugzilla.gnome.org/show_bug.cgi?id=752640
5779 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
5782 Automatic update of common submodule
5783 From f74b2df to 9aed1d7
5785 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
5790 === release 1.5.2 ===
5792 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
5798 * gst-rtsp-server.doap:
5801 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
5803 * gst/rtsp-server/rtsp-client.c:
5804 * gst/rtsp-server/rtsp-client.h:
5805 * tests/check/gst/client.c:
5806 rtsp-client: allow application to decide what requirements are supported
5807 Add "check-requirements" signal and vfunc to allow application
5808 (and subclasses) to check the requirements.
5809 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
5810 https://bugzilla.gnome.org/show_bug.cgi?id=749417
5812 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5815 Automatic update of common submodule
5816 From 6015d26 to f74b2df
5818 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5820 * gst/rtsp-server/rtsp-media.c:
5821 rtsp-media: Always use real payloader when creating streams
5822 A bin that contains the real payloader might be used as payloader. In this
5823 case we have to get the real payloader for the various properties it provides.
5824 Example use cases for this are bins that payload some media and then have
5825 additional elements that add metadata or RTP extension headers to the stream.
5826 https://bugzilla.gnome.org/show_bug.cgi?id=750800
5828 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5830 * examples/test-netclock-client.c:
5831 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
5833 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
5835 * examples/test-netclock-client.c:
5836 * examples/test-netclock.c:
5837 test-netclock: Use new ntp-time-source property on rtpbin
5838 Select the clock time to be used as NTP time source. This allows proper
5839 synchronization between receivers, independent of sharing base times, and just
5840 requires them to use the same clock.
5842 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5844 * examples/test-netclock-client.c:
5845 * examples/test-netclock.c:
5846 test-netclock: Setting the same base time on sender and receiver is not necessary
5847 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
5849 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5851 * gst/rtsp-server/rtsp-stream.c:
5852 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
5853 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5855 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5857 * docs/libs/gst-rtsp-server.types:
5858 docs: add missing types
5859 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5861 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5863 * docs/libs/gst-rtsp-server-sections.txt:
5864 docs: add missing apis
5865 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5867 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5869 * examples/test-netclock-client.c:
5870 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
5872 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5874 * docs/libs/gst-rtsp-server-sections.txt:
5875 * gst/rtsp-server/rtsp-auth.c:
5876 * gst/rtsp-server/rtsp-auth.h:
5877 GstRTSPAuth: Add client certificate authentication support
5878 https://bugzilla.gnome.org/show_bug.cgi?id=750471
5880 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5882 * examples/test-netclock-client.c:
5883 test-netclock-client: Use new GstClock API to wait for clock synchronization
5885 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
5887 * examples/test-netclock-client.c:
5888 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
5889 A mainloop is needed to get glimagesink to display something on OSX, and
5890 the source-setup signal just makes things a little bit easier.
5892 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
5895 Automatic update of common submodule
5896 From d9a3353 to 6015d26
5898 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
5901 Automatic update of common submodule
5902 From d37af32 to d9a3353
5904 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
5907 Automatic update of common submodule
5908 From 21ba2e5 to d37af32
5910 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
5913 Automatic update of common submodule
5914 From c408583 to 21ba2e5
5916 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
5918 * docs/libs/Makefile.am:
5919 docs: remove variables that we define in the snippet from common
5920 This is syncing our Makefile.am with upstream gtkdoc.
5922 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5925 Automatic update of common submodule
5926 From 44a3517 to c408583
5928 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
5933 === release 1.5.1 ===
5935 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
5941 * gst-rtsp-server.doap:
5944 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
5946 * gst/rtsp-server/rtsp-client.c:
5947 rtsp-client: No flush during Teardown.
5948 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
5949 backlog is empty it can happen that just a part of a message will be
5950 sent and rest is in backlog queue. If then flush during teardown
5951 just a part of message will be sent.This can lead to client miss
5952 teardown response since it expect to get the last part of message.
5953 The flushing during teardown was introduced to fix a deadlock that now
5954 is fixed more generally in handle_request by temporary setting backlog
5956 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
5958 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
5960 * tests/check/Makefile.am:
5961 tests: Use AM_TESTS_ENVIRONMENT
5962 Needed by the new automake test runner and the
5963 current version of the common submodule.
5965 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5967 * gst/rtsp-server/rtsp-media.h:
5968 * gst/rtsp-server/rtsp-stream.h:
5969 rtsp-server: Use single-include rtsp header to make sure we get all definitions
5971 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
5973 * gst/rtsp-server/rtsp-media.c:
5974 rtsp-media: Mark some more functions static
5976 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
5978 * gst/rtsp-server/rtsp-media.c:
5979 rtsp-media: Only unblock the media in suspend() when actually changing the state
5980 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
5982 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
5984 * examples/test-video-rtx.c:
5985 examples: Use AVPF profile for the RTX example
5987 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
5989 * gst/rtsp-server/rtsp-sdp.c:
5990 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
5992 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5994 * gst/rtsp-server/rtsp-stream.c:
5995 rtsp-stream: get valid clock-rate from last-sample
5996 clock-rate in last-sample's caps is integer, not unsigned.
5997 To get this value properly, variable needs to be type-casted to int.
5998 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6000 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6004 autogen.sh: only run autopoint if gettext requested in configure.ac
6005 Not just because there happens to be a po directory.
6006 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6008 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6011 Revert "configure.ac: uncomment gettext version setup"
6012 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6013 We don't need a gettext setup here and there's no po
6014 directory either, so no reason why autopoint would be
6015 run in the first place.
6016 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6018 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6020 * examples/test-multicast.c:
6021 * examples/test-multicast2.c:
6022 * examples/test-sdp.c:
6023 * examples/test-video-rtx.c:
6024 * examples/test-video.c:
6025 * tests/test-cleanup.c:
6026 * tests/test-reuse.c:
6027 Fix timeout function signatures across tests and examples
6029 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6031 * tests/check/Makefile.am:
6032 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6033 Make sure the test environment is set up.
6034 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6036 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6039 configure: bump automake requirement to 1.14 and autoconf to 2.69
6040 This is only required for builds from git, people can still
6041 build tarballs if they only have older autotools.
6042 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6044 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6047 configure.ac: uncomment gettext version setup
6048 Fixes autogen.sh. It would run autopoint, which would complain
6049 that it could not find the gettext version in configure.ac.
6050 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6052 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6054 * examples/test-video-rtx.c:
6055 test-video-rtx: set exact payload type to PCMA payloader
6056 Setting wrong payload type causes failure to do retransmission through audio stream
6057 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6059 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6061 * gst/rtsp-server/rtsp-media.c:
6062 * gst/rtsp-server/rtsp-stream.c:
6063 * gst/rtsp-server/rtsp-stream.h:
6064 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6065 Because of duplicated g_signal_connect for request-aux-sender signal,
6066 wrong stream pointer is passed to the signal handler.
6067 Instead of passing each stream, pass stream array and get the relevant stream.
6068 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6070 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6074 Update autogen.sh to latest version from common
6075 Fixes build after aclocal_check etc. helpers have been removed.
6077 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6080 Automatic update of common submodule
6081 From bc76a8b to c8fb372
6083 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6085 * gst/rtsp-server/rtsp-stream.c:
6086 rtsp-stream: Limit the queues to 1 buffer
6087 We only need them to be able to pre-roll, queueing up more data here
6088 is only going to harm latency and memory usage.
6090 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6092 * gst/rtsp-server/rtsp-stream.c:
6093 rtsp-stream: Update comment and ASCII art to the latest code
6094 We have a queue in front of the udpsink too to prevent the pipeline from
6097 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6099 * gst/rtsp-server/rtsp-stream.c:
6100 rtsp-media: Properly return first rtptime
6101 Instead we where returning first GstBuffer timestamp. This would result
6102 in clock skew and unwanted behaviour in RTSP playback.
6103 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6105 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6107 * gst/rtsp-server/rtsp-stream.c:
6108 rtsp-stream: Don't leave buffer mapped
6109 If the seq is NULL, the RTP buffer was left mapped. We should always
6112 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6117 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6119 * gst/rtsp-server/rtsp-media-factory.c:
6120 * tests/check/gst/client.c:
6121 Fix double semicolons
6123 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6125 * gst/rtsp-server/rtsp-stream.c:
6126 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6127 This gives more accurate values than asking the payloader. There might be
6128 queueing happening between the payloader and the sink.
6129 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6131 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6133 * gst/rtsp-server/rtsp-media.c:
6134 rtsp-media: Don't seek for PLAY if the position will not change
6135 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6137 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6139 * gst/rtsp-server/rtsp-media.c:
6140 rtsp-media: Don't include payload type in the caps for framesize
6141 When the sdp media attribute framesize are converted to caps
6142 the <payload> should not be included.
6143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6144 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6146 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6148 * gst/rtsp-server/rtsp-sdp.c:
6149 rtsp-sdp: add payload type to the sdp framesize attribute
6150 The sdp framesize attribute is desribed in RFC6064. It is specified
6151 for payloading of H263 and has the following form
6152 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6153 should be added to the caps in a payloader and the <payload type> should
6154 be added by the rtsp-server.
6155 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6157 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6159 * examples/test-uri.c:
6160 examples: test-uri: fix tainted variable
6161 Insignificant but this keeps Coverity happy.
6164 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6166 * examples/.gitignore:
6167 * examples/Makefile.am:
6168 * examples/test-netclock-client.c:
6169 * examples/test-netclock.c:
6170 examples: Add a simple example of network synch for live streams.
6171 An example server and client that works for synchronising live streams
6172 only - as it can't support pause/play.
6174 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6176 * gst/rtsp-server/rtsp-media-factory.c:
6177 * gst/rtsp-server/rtsp-media-factory.h:
6178 rtsp-media-factory: Add functions to set/get the media gtype
6179 Allow specifying the GType of a GstRtspMedia subclass to create
6180 as a simpler way to get the factory to create a custom
6181 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6183 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6185 * gst/rtsp-server/rtsp-media.c:
6186 rtsp-media: fix double unlock in _get_buffer_size()
6187 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6188 because of double g_mutex_unlock () usage.
6189 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6191 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6193 * gst/rtsp-server/rtsp-session-pool.c:
6194 * gst/rtsp-server/rtsp-session.c:
6195 * gst/rtsp-server/rtsp-session.h:
6196 rtsp-session: Use monotonic time for RTSP session timeout
6197 Changed RTSP session timeout handling to monotonic time
6198 and deprecating the API for current system time.
6199 This fixes timeouts when the system time changes.
6200 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6202 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6204 * gst/rtsp-server/rtsp-client.c:
6205 * gst/rtsp-server/rtsp-media.c:
6206 rtsp-client: Only error out in PLAY if seeking actually failed
6207 If the media was just not seekable, we continue from whatever position we are
6208 and let the client decide if that is what is wanted or not.
6209 Only if the actual seek failed, we can't really recover and should error out.
6211 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6213 * gst/rtsp-server/rtsp-stream.c:
6214 rtsp-stream: Add necessary queues between tee and multiudpsink
6215 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6217 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6219 * gst/rtsp-server/rtsp-client.c:
6220 * gst/rtsp-server/rtsp-media.c:
6221 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6222 Instead error out properly the same way as if the SEEKING query already
6225 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6227 * gst/rtsp-server/rtsp-stream.h:
6228 rtsp-stream: minor code formatting fix
6230 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6232 * gst/rtsp-server/rtsp-media.c:
6233 rtsp-media: fix logic for collect_streams
6234 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6235 all streams it knows if it got any, and can check if the transport mode is OK.
6238 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6240 * gst/rtsp-server/rtsp-media.c:
6241 rtsp-media: Don't set the transport mode based on what elements we find
6242 Just print a warning if the one that was set before disagrees with what
6243 elements we found. It must already be set to something before as this
6244 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6245 and we would reject ANNOUNCE if the RECORD flag was not set.
6247 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6249 * tests/check/gst/rtspserver.c:
6250 tests: rtspserver: rename shadowed variable
6251 We have two different 'sink' variables here,
6252 rename one of them for clarity.
6254 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6256 * gst/rtsp-server/rtsp-client.c:
6257 rtsp-client: fix awkward if clause
6259 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6261 * examples/test-uri.c:
6262 examples: test-uri: improve uri argument handling and accept file names
6263 Print an error if the argument passed is not a URI and can't
6264 be converted into one, or no arguments have been provided.
6266 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6268 * examples/test-uri.c:
6269 examples: test-uri: don't remove mount point after 10 seconds
6270 It's very irritating when trying to test stuff repeatedly
6271 and serves no real purpose other than showing that it can
6274 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6276 * examples/.gitignore:
6277 examples: add new test-record to .gitignore
6279 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6281 * examples/test-record.c:
6282 * gst/rtsp-server/rtsp-client.c:
6283 * gst/rtsp-server/rtsp-media-factory.c:
6284 * gst/rtsp-server/rtsp-media-factory.h:
6285 * gst/rtsp-server/rtsp-media.c:
6286 * gst/rtsp-server/rtsp-media.h:
6287 * tests/check/gst/rtspserver.c:
6288 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6290 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6292 * examples/test-record.c:
6293 test-record: Set latency for playback-style example to 2s instead of 200ms
6295 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6297 * tests/check/gst/rtspserver.c:
6298 tests: add some unit tests for ANNOUNCE and RECORD
6299 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6301 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6303 * gst/rtsp-server/rtsp-client.c:
6304 rtsp-client: fix a couple of leaks in handle_announce
6306 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6308 * gst/rtsp-server/rtsp-media-factory.c:
6309 * gst/rtsp-server/rtsp-media-factory.h:
6310 * gst/rtsp-server/rtsp-media.c:
6311 * gst/rtsp-server/rtsp-media.h:
6312 rtsp-media: Expose latency setting for setting the rtpbin latency
6314 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6316 * examples/test-record.c:
6317 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6319 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6321 * gst/rtsp-server/rtsp-stream.c:
6322 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6324 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6326 * examples/Makefile.am:
6327 * examples/test-record.c:
6328 * gst/rtsp-server/rtsp-client.c:
6329 * gst/rtsp-server/rtsp-client.h:
6330 * gst/rtsp-server/rtsp-media-factory.c:
6331 * gst/rtsp-server/rtsp-media-factory.h:
6332 * gst/rtsp-server/rtsp-media.c:
6333 * gst/rtsp-server/rtsp-media.h:
6334 * gst/rtsp-server/rtsp-session-media.c:
6335 * gst/rtsp-server/rtsp-stream.c:
6336 * gst/rtsp-server/rtsp-stream.h:
6337 Add initial support for RECORD
6338 We currently only support media that is RECORD or PLAY only, not both at once.
6339 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6341 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6343 * gst/rtsp-server/rtsp-stream.c:
6344 rtsp-stream: RTCP and RTP transport cache cookies seperated
6345 RTCP packets were not sent because the same tr_cache_cookie was used for
6346 both RTP and RTCP. So only one of the tr_cache lists were populated
6347 depending on which one was sent first. If the tr_cache list is not
6348 populated then no packets can be sent. Most often this happened to be
6349 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6350 resulted in both the tr_cache_lists to be populated regardless of which
6352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6354 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6356 * gst/rtsp-server/rtsp-stream.c:
6357 rtsp-stream: fix false compiler warning
6358 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6360 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6362 * gst/rtsp-server/rtsp-client.c:
6363 rtsp-client: log interleaved data received
6365 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6367 * gst/rtsp-server/rtsp-client.c:
6368 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6370 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6372 * gst/rtsp-server/rtsp-client.c:
6373 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6375 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6377 * gst/rtsp-server/rtsp-client.c:
6378 rtsp-client: Use a random session ID in the SDP
6379 RFC4566 Section 5.2 says that it should make the username, session id,
6380 nettype, addrtype and unicast address tuple globally unique. Always using
6381 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6382 Instead let's create a 64 bit random number, which at least brings us
6383 closer to the goal of global uniqueness.
6384 https://tools.ietf.org/html/rfc4566#section-5.2
6386 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6388 * examples/test-launch.c:
6389 * examples/test-mp4.c:
6390 * examples/test-ogg.c:
6391 * examples/test-uri.c:
6392 examples: Don't call gst_init() and gst_get_option_group()
6393 The latter calls the former at the appropriate time.
6395 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6397 * gst/rtsp-server/rtsp-client.c:
6398 rtsp-client: Drop trailing \0 of RTSP DATA messages
6399 We add a trailing \0 in GstRTSPConnection to make parsing of
6400 string message bodies easier (e.g. the SDP from DESCRIBE) but
6401 for actual data this means we have to drop it or otherwise
6402 create invalid data.
6404 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6406 * gst/rtsp-server/rtsp-stream.c:
6407 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6408 Fixes crash when two threads access handle_new_sample() at the same
6409 time, one for RTP, one for RTCP.
6410 Otherwise, when iterating over the transports cache, it might be modified by
6411 another thread at the same time if the transports cookie has changed.
6412 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6414 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6416 * gst/rtsp-server/rtsp-stream.c:
6417 rtsp-stream: Set format=TIME on our app sources for TCP
6419 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6421 * gst/rtsp-server/rtsp-session-pool.c:
6422 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6423 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6424 RFC 2326 states that session IDs may consist of alphanumeric as well as
6425 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6426 Previously the session ID was URI-escaped, this meant that any character
6427 which was not alphanumeric or any of the characters +-._~ would be
6428 percent encoded. While the RFC (surprisingly) mentions that linear white
6429 space in session IDs should be URI-escaped, it does not say anything
6430 about other characters. Moreover no white space is allowed in the
6431 session ID. Finally the percent character which is the result of
6432 URI-escaping is not allowed in a session ID.
6433 So there is no reason to do any URI-escaping, and now it is removed.
6434 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6436 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6439 Automatic update of common submodule
6440 From f2c6b95 to bc76a8b
6442 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6445 Fix 'make check' from top-level directory
6447 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6449 * examples/test-launch.c:
6450 * examples/test-mp4.c:
6451 * examples/test-ogg.c:
6452 * examples/test-uri.c:
6453 examples: Add command-line parsing and take a 'port' argument
6454 This allows users to run multiple servers on different ports for testing.
6455 Only done for examples that actually take arguments and hence are capable of
6456 outputting different streams for each instance on each port.
6457 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6459 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6461 * gst/rtsp-server/rtsp-client.c:
6462 * gst/rtsp-server/rtsp-client.h:
6463 rtsp-client: Add a send_message default signal handler
6464 This allows subclasses to easily hook into the response sending
6465 mechanism without doing everything from a signal, which seems
6466 awkward from subclasses.
6468 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6471 Automatic update of common submodule
6472 From ef1ffdc to f2c6b95
6474 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6478 configure: add --disable-examples switch
6479 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6481 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6483 * examples/.gitignore:
6484 * examples/Makefile.am:
6485 * examples/test-video-rtx.c:
6486 examples: add a retransmisison example implementing RFC4588
6487 Currently only SSRC-multiplexed rtx streams are supported
6489 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6491 * gst/rtsp-server/rtsp-stream.c:
6492 rtsp-stream: Fix some minor memory leaks
6494 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6496 * gst/rtsp-server/rtsp-media.c:
6497 rtsp-media: Some minor cleanup
6499 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6501 * gst/rtsp-server/rtsp-stream.c:
6502 rtsp-stream: Fix compiler warnings
6503 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6504 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6506 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6507 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6510 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6512 * docs/libs/gst-rtsp-server-sections.txt:
6513 * gst/rtsp-server/rtsp-media-factory.c:
6514 * gst/rtsp-server/rtsp-media-factory.h:
6515 * gst/rtsp-server/rtsp-media.c:
6516 * gst/rtsp-server/rtsp-media.h:
6517 * gst/rtsp-server/rtsp-sdp.c:
6518 * gst/rtsp-server/rtsp-stream.c:
6519 * gst/rtsp-server/rtsp-stream.h:
6520 media: implement ssrc-multiplexed retransmission support
6521 based off RFC 4588 and the server-rtpaux example in -good
6523 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6525 * gst/rtsp-server/rtsp-client.c:
6526 * gst/rtsp-server/rtsp-stream-transport.c:
6527 * gst/rtsp-server/rtsp-stream.c:
6528 rtsp: Ref transports in hash table.
6529 Also ref streams for transports.
6530 This solves a crash when reciving a rtcp after teardown but before
6532 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6534 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6537 Automatic update of common submodule
6538 From 7bb2bce to ef1ffdc
6540 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6542 * gst/rtsp-server/rtsp-client.c:
6543 client: refactor cleanup of cached media
6545 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6547 * tests/check/gst/client.c:
6549 The session leak is now fixed, lets remove those FIXME comments.
6551 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6553 * tests/check/gst/rtspserver.c:
6554 tests: Test to setup two sessions on one connection
6555 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6557 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6559 * tests/check/gst/rtspserver.c:
6560 tests: Test setup with tcp transport
6561 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6563 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6565 * gst/rtsp-server/rtsp-client.c:
6566 client: Configure transport after creating session media
6567 The default implementation of configure_client_transport() in
6568 rtsp-client uses the session media when it chooses channels for
6569 interleaved traffic.
6570 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6572 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6574 * gst/rtsp-server/rtsp-client.c:
6575 * gst/rtsp-server/rtsp-session-media.c:
6576 client: Stop caching media in client when doing setup
6577 If the media has been managed by a session media, it should not be
6578 cached in the client any longer. The GstRTSPSessionMedia object is now
6579 responsible for unpreparing the GstRTSPMedia object using
6580 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6582 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6584 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6586 * gst/rtsp-server/rtsp-stream.c:
6587 rtsp-stream: unref srtp decoder when leaving bin
6588 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6590 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6592 * gst/rtsp-server/rtsp-client.c:
6593 rtsp-client: mikey memory leaks
6594 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6596 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6599 Automatic update of common submodule
6600 From 84d06cd to 7bb2bce
6602 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6605 Parallelise 'make check-valgrind'
6607 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6610 Automatic update of common submodule
6611 From a8c8939 to 84d06cd
6613 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6616 Automatic update of common submodule
6617 From 36388a1 to a8c8939
6619 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6621 * gst/rtsp-server/rtsp-media.c:
6622 rtsp-media: deactivate media when shutting down from paused
6623 This was only done when going directly from playing.
6624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
6626 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6628 * gst/rtsp-server/rtsp-client.c:
6629 * gst/rtsp-server/rtsp-context.h:
6630 rtsp-client: add stream transport to context
6631 We add the stream transport to the context so we can get the configured
6632 client stream transport in the setup request signal.
6633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
6635 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6637 * gst/rtsp-server/rtsp-stream.c:
6638 stream: release lock even not all transports have been removed
6639 We don't want to keep the lock even we return FALSE because not all the
6640 transports have been removed. This could lead into a deadlock.
6641 https://bugzilla.gnome.org/show_bug.cgi?id=737797
6643 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
6645 * gst/rtsp-server/rtsp-sdp.c:
6646 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
6647 These were renamed in GstRTPBasePayload in 1.0
6649 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6651 * gst/rtsp-server/rtsp-client.c:
6652 client: set session media to NULL without the lock
6653 We need to set session medias to NULL without the client lock otherwise
6654 we can end up in a deadlock if another thread is waiting for the lock
6655 and media unprepare is also waiting for that thread to end.
6656 https://bugzilla.gnome.org/show_bug.cgi?id=737690
6658 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
6660 * gst/rtsp-server/rtsp-media.c:
6661 rtsp-media: Set state to UNPREPARING in all cases
6663 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
6665 * gst/rtsp-server/rtsp-media.c:
6666 media: set state to unpreparing when unprepare is initiated
6667 https://bugzilla.gnome.org/show_bug.cgi?id=737675
6669 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
6671 * gst/rtsp-server/rtsp-client.c:
6672 rtsp-client: Remove backlog limit while processings requests
6673 If the backlog limit is kept two cases of deadlocks may be
6674 encountered when streaming over TCP. Without the backlog
6675 limit this deadlocks can not happen, at the expence of
6677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
6679 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
6681 * gst/rtsp-server/rtsp-client.c:
6682 rtsp-client: do not free main context before rtsp watch
6683 https://bugzilla.gnome.org/show_bug.cgi?id=737110
6685 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
6687 * tests/check/gst/rtspserver.c:
6688 tests: Extend unit test timeout to accomodate for valgrind
6689 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6691 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
6693 * gst/rtsp-server/rtsp-client.c:
6694 * gst/rtsp-server/rtsp-session.c:
6695 * gst/rtsp-server/rtsp-stream-transport.c:
6696 rtsp-*: Treat sending packets to clients as keepalive
6697 As long as gst-rtsp-server can successfully send RTP/RTCP data to
6698 clients then the client must be reading. This change makes the server
6699 timeout the connection if the client stops reading.
6700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6702 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
6704 * gst/rtsp-server/rtsp-client.c:
6705 rtsp-client: Allow backlog to grow while expiring session
6706 Allow the send backlog in the RTSP watch to grow to unlimited size while
6707 attempting to bring the media pipeline to NULL due to a session
6708 expiring. Without this change the appsink element cannot change state
6709 because it is blocked while rendering data in the new_sample callback.
6710 This callback will block until it has successfully put the data into the
6711 send backlog. There is a chance that the send backlog is full at this
6712 point which means that the callback may block for a long time, possibly
6713 forever. Therefore the media pipeline may also be prevented from
6714 changing state for a long time.
6715 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6717 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
6719 * gst/rtsp-server/rtsp-client.c:
6720 rtsp-client: Make old compilers happy
6721 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
6722 Just in case that guint8 doesn't fit in a pointer. Just in case ...
6724 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
6726 * gst/rtsp-server/rtsp-client.c:
6727 client: raise the backlog limits before pausing
6728 We need to raise the backlog limits before pausing the pipeline or else
6729 the appsink might be blocking in the render method in wait_backlog() and
6730 we would deadlock waiting for paused.
6731 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
6733 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
6735 * gst/rtsp-server/rtsp-client.c:
6736 client: make define for the WATCH_BACKLOG
6737 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
6739 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
6741 * gst/rtsp-server/rtsp-client.c:
6742 client: simplify session transport handling
6743 link/unlink of the transport in a session was done to keep track of all
6744 TCP transports and to send RTP/RTCP data to the streams. We can simplify
6745 that by putting all the TCP transports in a hashtable indexed with the
6747 We also don't need to link/unlink the transports when we pause/resume
6748 the streams. The same effect is already achieved when we pause/play the
6749 media. Indeed, when we pause the media, the transport is removed from
6750 the media and the callbacks will not be called anymore.
6751 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
6753 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
6755 * gst/rtsp-server/rtsp-stream-transport.c:
6756 * gst/rtsp-server/rtsp-stream-transport.h:
6757 stream-transport: make method to handle received data
6758 Make a method to handle the data received on a channel. It sends the
6759 data to the stream of the transport on the RTP or RTCP pads based on
6762 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
6764 * examples/test-mp4.c:
6765 test: add example of dumping RTCP reports
6767 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
6769 * gst/rtsp-server/rtsp-media.c:
6770 * gst/rtsp-server/rtsp-stream.c:
6771 * gst/rtsp-server/rtsp-stream.h:
6772 rtsp-media: Make sure that sequence numbers are monotonic after pause
6773 The sequence number is not monotonic for RTP packets after pause. The
6774 reason is basepayloader generates a randon sequence number when the
6775 pipeline goes from ready to pause. With this fix generation of sequence
6776 number will be monotonic when going from pause to play request.
6777 https://bugzilla.gnome.org/show_bug.cgi?id=736017
6779 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
6781 * gst/rtsp-server/rtsp-client.c:
6782 rtsp-client: Protect saved clients watch with a mutex
6783 Fixes a crash when close() is called while merging clients
6784 in handle_tunnel(). In that case close() would destroy the
6785 watch while it is still being used in handle_tunnel().
6786 https://bugzilla.gnome.org/show_bug.cgi?id=735570
6788 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
6790 * gst/rtsp-server/rtsp-stream.c:
6791 rtsp-stream: Remove the multicast group udp sources when removing from the bin
6793 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6795 * gst/rtsp-server/rtsp-media.c:
6796 * gst/rtsp-server/rtsp-stream.c:
6797 * gst/rtsp-server/rtsp-stream.h:
6798 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
6799 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
6800 seeking and will always continue counting the time. This leads to
6801 the NPT after a backwards seek to be something completely different
6802 to the actual seek position.
6803 https://bugzilla.gnome.org/show_bug.cgi?id=732644
6805 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
6807 * examples/test-appsrc.c:
6808 examples: fix another reference leak
6809 gst_rtsp_media_get_element() returns a new ref.
6811 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6813 * examples/test-appsrc.c:
6814 examples: unref element after usage
6815 gst_bin_get_by_name_recurse_up() returns an element
6816 reference that must be unreffed after usage.
6817 https://bugzilla.gnome.org/show_bug.cgi?id=734546
6819 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
6821 * gst/rtsp-server/rtsp-media.c:
6822 signals: Fix copy-pasto in target-state signal offset
6824 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
6828 Makefile: Add usage of build-checks step
6829 Allows building checks without running them
6831 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
6833 * gst/rtsp-server/rtsp-stream.c:
6834 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
6835 When a UDP multicast transport is used it is expected that the server listens
6836 for RTP and RTCP packets on the multicast group with the corresponding port.
6837 Without this we will never get RTCP packets from clients in multicast mode.
6838 https://bugzilla.gnome.org/show_bug.cgi?id=732238
6840 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6845 === release 1.4.0 ===
6847 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6853 * gst-rtsp-server.doap:
6856 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
6858 * gst/rtsp-server/rtsp-media.h:
6859 media: correct misspelled words in description
6860 https://bugzilla.gnome.org/show_bug.cgi?id=733244
6862 === release 1.3.91 ===
6864 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6870 * gst-rtsp-server.doap:
6873 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
6875 * docs/libs/gst-rtsp-server-sections.txt:
6878 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
6880 * gst/rtsp-server/rtsp-server.c:
6881 server: implement client REMOVE filter
6883 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
6885 * gst/rtsp-server/rtsp-client.c:
6886 * gst/rtsp-server/rtsp-client.h:
6887 client: expose _close() method
6888 Expose a previously internal close method to close the client
6891 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
6893 * gst/rtsp-server/rtsp-session-pool.c:
6894 session-pool: signal session-removed outside of the lock
6895 Release the lock before emiting the session-removed signal.
6897 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
6899 * gst/rtsp-server/rtsp-client.c:
6900 * gst/rtsp-server/rtsp-server.c:
6901 * gst/rtsp-server/rtsp-session-pool.c:
6902 * gst/rtsp-server/rtsp-session.c:
6903 * gst/rtsp-server/rtsp-stream.c:
6904 filter: Release lock in filter functions
6905 Release the object lock before calling the filter functions. We need to
6906 keep a cookie to detect when the list changed during the filter
6907 callback. We also keep a hashtable to make sure we only call the filter
6908 function once for each object in case of concurrent modification.
6909 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
6911 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
6913 * gst/rtsp-server/rtsp-client.c:
6914 client: check if watch is set in handle_teardown()
6915 The unit tests run without a watch
6917 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6919 * tests/check/gst/client.c:
6920 client tests: send teardown to cleanup session
6922 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
6924 * tests/check/gst/rtspserver.c:
6925 server tests: send teardown to cleanup session
6927 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6929 * gst/rtsp-server/rtsp-client.c:
6930 client: keep ref to client for the session removed handler
6931 This extra ref will be dropped when all client sessions have been
6932 removed. A session is removed when a client sends teardown, closes its
6933 endpoint of the TCP connection or the sessions expires.
6934 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6936 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
6938 * gst/rtsp-server/rtsp-client.c:
6939 * gst/rtsp-server/rtsp-session.c:
6940 * tests/check/gst/client.c:
6941 client: manage media in session as a last step
6942 Once we manage a media in a session, we can't unmanage it anymore
6943 without destroying it. Therefore, first check everything before we
6944 manage the media, otherwise if something is wrong we have no way to
6946 If we created a new session and something went wrong, remove the session
6947 again. Fixes a leak in the unit test.
6949 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
6951 * examples/test-mp4.c:
6952 * examples/test-ogg.c:
6953 examples: print 'stream ready at url' for mp4 and ogg example
6955 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
6957 * gst/rtsp-server/rtsp-client.c:
6958 * gst/rtsp-server/rtsp-sdp.c:
6959 rtsp: fix for MIKEY api change
6961 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
6963 * gst/rtsp-server/rtsp-client.c:
6964 client: free watch context only once
6965 The watch context is freed when the source is destroyed. Avoids
6966 a CRITICAL when we try to unref the context twice.
6968 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
6970 * gst/rtsp-server/rtsp-client.c:
6973 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
6975 * gst/rtsp-server/rtsp-client.c:
6976 client: protect sessions with lock
6977 Protect the list of sessions with the lock.
6978 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6980 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
6982 * gst/rtsp-server/rtsp-client.c:
6983 Client: keep a ref to the session
6984 Don't just keep a weak ref to the session objects but use a hard ref. We
6985 will be notified when a session is removed from the pool (expired) with
6986 the new session-removed signal.
6987 Don't automatically close the RTSP connection when all the sessions of
6988 a client are removed, a client can continue to operate and it can create
6989 a new session if it wants. If you want to remove the client from the
6990 server, you have to use gst_rtsp_server_client_filter() now.
6991 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
6992 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
6994 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
6996 * gst/rtsp-server/rtsp-session-pool.c:
6997 * gst/rtsp-server/rtsp-session-pool.h:
6998 session-pool: add session-removed signal
6999 Add a signal to be notified when a session is removed from the pool.
7001 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7003 * gst/rtsp-server/Makefile.am:
7004 * gst/rtsp-server/rtsp-server.h:
7005 Make rtsp-server.h a single-include header, use it for G-I
7006 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7008 === release 1.3.90 ===
7010 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7016 * gst-rtsp-server.doap:
7019 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7021 * gst/rtsp-server/rtsp-stream.c:
7022 stream: crypto can be NULL
7024 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7026 * gst/rtsp-server/rtsp-client.c:
7027 * gst/rtsp-server/rtsp-media.c:
7028 * gst/rtsp-server/rtsp-mount-points.c:
7029 introspection: add missing allow-none annotations
7030 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7032 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7034 * gst/rtsp-server/rtsp-address-pool.c:
7035 * gst/rtsp-server/rtsp-media.c:
7036 * gst/rtsp-server/rtsp-session-media.c:
7037 * gst/rtsp-server/rtsp-session-pool.c:
7038 * gst/rtsp-server/rtsp-stream-transport.c:
7039 * gst/rtsp-server/rtsp-stream.c:
7040 * gst/rtsp-server/rtsp-token.c:
7041 introspection: add (nullable) annotations to return values
7042 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7044 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7046 * gst/rtsp-server/rtsp-client.c:
7047 * gst/rtsp-server/rtsp-stream.c:
7048 gi: improve annotations
7049 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7051 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7053 * gst/rtsp-server/rtsp-client.c:
7054 * gst/rtsp-server/rtsp-media-factory.c:
7055 * gst/rtsp-server/rtsp-media.c:
7056 * gst/rtsp-server/rtsp-server.c:
7057 signals: use generic marshal function
7058 Use the generic C marshal function.
7059 Use more explicit type instead of G_TYPE_POINTER
7061 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7063 * gst/rtsp-server/rtsp-context.h:
7064 context: add type macro
7066 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7068 * gst/rtsp-server/rtsp-client.c:
7069 * gst/rtsp-server/rtsp-sdp.c:
7070 * gst/rtsp-server/rtsp-sdp.h:
7071 sdp: hide key length defines
7072 They don't have a namespace.
7074 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7079 === release 1.3.3 ===
7081 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7087 * gst-rtsp-server.doap:
7090 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7092 * gst/rtsp-server/rtsp-client.c:
7093 * gst/rtsp-server/rtsp-sdp.c:
7094 * gst/rtsp-server/rtsp-sdp.h:
7095 mikey: add different key length parameters
7096 Add encryption and authentication key length parameters to MIKEY. For
7097 the encoders, the key lengths are obtained from the cipher and auth
7098 algorithms set in the caps. For the decoders, they are obtained while
7099 parsing the key management from the client.
7100 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7102 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7104 * tests/check/gst/stream.c:
7105 stream tests: Make sure we get right multicast address from stream
7106 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7108 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7110 * gst/rtsp-server/rtsp-client.c:
7111 client: ref the context until rtsp watch is alive
7112 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7114 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7116 * gst/rtsp-server/rtsp-client.c:
7117 client: Destroy the rtsp watch after connection close
7119 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7121 * gst/rtsp-server/rtsp-media.c:
7122 media: fix confusing comment
7124 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7126 * gst/rtsp-server/rtsp-session.c:
7127 rtsp-session: Timeout in header.
7128 Adding the possbilty to always have timout in header.
7129 This is configurabe with setting "timeout-always-visible".
7130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7132 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7137 === release 1.3.2 ===
7139 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7146 * gst-rtsp-server.doap:
7149 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7152 Automatic update of common submodule
7153 From 211fa5f to 1f5d3c3
7155 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7157 * gst/rtsp-server/rtsp-client.c:
7158 client: store TCP ports in transport
7159 Store the TCP ports in the transport when we are doing RTSP over TCP.
7160 This way, we can easily get to the ports from the transport.
7161 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7163 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7165 * gst/rtsp-server/rtsp-stream.c:
7166 stream: add signals for new RTP/RTCP encoders
7167 New signals to allow the user to configure the dynamically created
7169 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7171 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7173 * gst/rtsp-server/rtsp-media.c:
7174 * gst/rtsp-server/rtsp-media.h:
7175 media: Make suspend()/unsuspend() virtual
7176 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7178 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7180 * gst/rtsp-server/rtsp-client.c:
7181 client: fix send-message signal marshaller
7182 Use generic marshalling for the send-message signal. It has
7183 two POINTER arguments, not just one.
7184 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7186 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7188 * tests/check/gst/media.c:
7189 tests: add and remove pads only once
7190 In this test we simulate a dynamic pad by watching the caps event.
7191 Because of renegotiation in the base payloader now, this caps is sent
7192 multiple times but we can only deal with 1 invocation, use a variable to
7193 only 'add and remove' the pad once.
7195 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7197 * tests/check/gst/rtspserver.c:
7198 tests: add unit test for correct handling of Require headers
7199 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7201 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7203 * gst/rtsp-server/rtsp-client.c:
7204 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7205 Servers must handle Require headers and must report a failure
7206 if they don't handle any of the Required options, see RFC 2326,
7207 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7208 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7210 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7215 === release 1.3.1 ===
7217 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7223 * gst-rtsp-server.doap:
7226 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7229 Automatic update of common submodule
7230 From bcb1518 to 211fa5f
7232 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7237 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7239 * tests/check/gst/sessionmedia.c:
7240 tests: fix memory leak in sessionmedia unit test
7242 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7244 * gst/rtsp-server/rtsp-client.c:
7245 client: emit a signal before sending a message
7246 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7248 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7250 * gst/rtsp-server/rtsp-client.c:
7251 client: pass context to send_message
7252 Pass the current context to send_message, we will need it later.
7254 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7256 * gst/rtsp-server/rtsp-client.c:
7257 client: fix typo in comment
7259 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7261 * gst/rtsp-server/rtsp-media.c:
7262 media: Do not stop thread twice if default_prepare() fails
7264 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7266 * gst/rtsp-server/rtsp-client.c:
7267 client: set the watch to flushing before going to NULL
7268 First set the watch to flushing so that we unblock any current and
7269 future attempt to send data on the watch, Then set the pipeline to
7271 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7273 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7275 * gst/rtsp-server/rtsp-session-pool.c:
7276 * tests/check/gst/sessionpool.c:
7277 rtsp-session-pool: Fixes annotation
7278 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7279 in the sessionpool test.
7280 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7282 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7284 * gst/rtsp-server/rtsp-media.c:
7285 * gst/rtsp-server/rtsp-media.h:
7286 media: make media_prepare virtual
7287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7289 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7291 * gst/rtsp-server/rtsp-media.c:
7292 * tests/check/gst/media.c:
7293 media: stop the thread in more error cases
7295 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7297 * gst/rtsp-server/rtsp-media.c:
7298 * tests/check/gst/media.c:
7299 media: allow NULL as the thread
7300 Use the default context whan passing a NULL thread.
7302 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7304 * gst/rtsp-server/rtsp-client.c:
7305 rtsp-client: indent cleanup
7306 Coverity was moaning about unreachable code, and I think it was just
7307 confused by { being before the label. We'll see if it pops up again.
7310 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7312 * gst/rtsp-server/rtsp-client.c:
7313 * gst/rtsp-server/rtsp-media.c:
7314 client: Add drop-backlog property
7315 When we have too many messages queued for a client (currently hardcoded
7316 to 100) we overflow and drop the messages. Add a drop-backlog property
7317 to control this behaviour. Setting this property to FALSE will retry
7318 to send the messages to the client by waiting for more room in the
7320 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7322 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7324 * gst/rtsp-server/rtsp-client.c:
7325 client: support for POST before GET when setting up a tunnel
7327 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7329 * gst/rtsp-server/rtsp-client.c:
7330 client: remove watch of the second client after http tunnel setup
7331 The second client will be freed after the HTTP tunnel has been set up.
7332 Make sure it's RTSP watch is never dispatched again.
7333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7335 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7337 * gst/rtsp-server/rtsp-media.c:
7338 * tests/check/gst/media.c:
7339 media: Make media_prepare() fail if port allocation fails
7340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7342 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7344 * tests/check/gst/media.c:
7345 media test: cleanup the thread pool in tests
7347 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7349 * gst/rtsp-server/rtsp-media.c:
7350 * tests/check/gst/media.c:
7351 rtsp-media: Unblock blocked streams in unprepare
7352 The streams will be blocked when a live media is prepared.
7353 The streams should be unblocked in gst_rtsp_media_unprepare.
7354 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7356 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7358 * gst/rtsp-server/rtsp-media.c:
7359 media: release the state lock when going to NULL
7360 Set our state to UNPREPARING and release the state-lock before
7361 setting the pipeline to the NULL state. This way, any pad-added
7362 callback will be able to take the state-lock and check that we are now
7363 unpreparing instead of deadlocking.
7364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7366 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7368 * gst/rtsp-server/rtsp-media.c:
7369 media: protect status with lock
7370 Make sure we only update the status with the lock.
7372 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7374 * gst/rtsp-server/rtsp-client.c:
7375 * gst/rtsp-server/rtsp-sdp.c:
7376 rtsp: update for MIKEY API changes
7378 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7380 * gst/rtsp-server/rtsp-client.c:
7381 client: parse the mikey response from the client
7382 Parse the mikey response from the client and update the policy for
7385 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7387 * gst/rtsp-server/rtsp-stream.c:
7388 * gst/rtsp-server/rtsp-stream.h:
7389 stream: add method to set crypto info
7390 Make a method to configure the crypto information of a stream.
7391 Set udpsrc in READY instead of PAUSED so that we can configure caps
7394 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7396 * gst/rtsp-server/rtsp-client.c:
7397 client: cleanup error paths
7399 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7401 * gst/rtsp-server/rtsp-media.c:
7404 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7406 * examples/test-video.c:
7407 test: enable SRTP only on RTSPS
7408 We only want to enable SRTP when doing rtsp over TLS so that we can
7409 exchange the keys in a secure way.
7411 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7413 * examples/test-video.c:
7414 test: print an error on failure
7416 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7419 * examples/test-video.c:
7420 * gst/rtsp-server/rtsp-sdp.c:
7421 * gst/rtsp-server/rtsp-stream.c:
7422 * tests/check/Makefile.am:
7423 stream: add SRTP support
7424 Install srtp encoder and decoder elements in rtpbin
7427 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7429 * tests/check/Makefile.am:
7430 * tests/check/gst/sessionpool.c:
7431 tests: Add unit tests for sessionpool
7432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7434 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7436 * tests/check/gst/threadpool.c:
7437 tests: Improve code coverage of rtsp-threadpool tests
7438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7440 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7442 * tests/check/gst/sessionmedia.c:
7443 tests: Improve code coverage for rtsp-session-media
7444 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7446 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7448 gobject-introspection: Add annotations to support language bindings
7449 In addition a few cosmetic changes:
7450 * Adjust the order of arguments
7451 * Fix typo: occured -> occurred
7452 * Fix indentation after Return:-clauses
7453 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7455 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7457 * gst/rtsp-server/rtsp-stream.c:
7458 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7459 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7461 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7463 * gst/rtsp-server/rtsp-stream.c:
7464 stream: take caps after the session manager
7465 Take the caps for the SDP after they leave the rtpbin so that we can
7466 also get the properties added by rtpbin elements.
7468 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7470 * gst/rtsp-server/rtsp-stream.c:
7471 stream: release lock while pushing out packets
7472 Keep a cache of the transports and use this to iterate the transport
7473 while pushing packets. This allows us to release the lock early.
7474 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7476 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7478 * gst/rtsp-server/rtsp-client.c:
7479 * gst/rtsp-server/rtsp-client.h:
7480 rtsp-client: vmethod for modifying tunnel GET response
7481 Add a vmethod tunnel_http_response where the response to the HTTP GET
7482 for tunneled connections can be modified.
7483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7485 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7487 * gst/rtsp-server/rtsp-sdp.c:
7488 sdp: make 1 media line per profile
7489 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7490 line in the SDP for each profile. The client is then supposed to pick
7491 one of the profiles in the SETUP request. Because the m= lines have the
7492 same pt, the client also knows that only 1 option is possible.
7494 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7496 * gst/rtsp-server/rtsp-media-factory.c:
7497 * gst/rtsp-server/rtsp-media-factory.h:
7498 * gst/rtsp-server/rtsp-media.c:
7499 factory: add profile property and pass to media and streams
7501 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7503 * examples/test-multicast.c:
7504 * gst/rtsp-server/rtsp-sdp.c:
7505 sdp: pass multicast connection for multicast-only stream
7506 Pass the multicast address of the stream in the connection info in the
7507 SDP so that clients try a multicast connection first.
7508 Only allow multicast connections in the test-multicast example. Also
7509 increase the TTL a little.
7511 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7514 .gitignore: Ignore gcov intermediate files
7515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7517 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7519 * gst/rtsp-server/rtsp-stream.c:
7520 stream: release some locks in error cases
7522 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7524 docs: Enable and fix gtk-doc warnings
7525 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7526 * addresspool/mediafactory: Add missing annotation colon
7527 * stream: Annotate return value
7528 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7530 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7533 Automatic update of common submodule
7534 From fe1672e to bcb1518
7536 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7539 Automatic update of common submodule
7540 From 1a07da9 to fe1672e
7542 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7544 * examples/Makefile.am:
7545 examples: use LDADD for libs instead of LDFLAGS
7547 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7550 configure: make sure releases are in .doap file
7552 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7554 * examples/test-cgroups.c:
7555 examples: test-cgroups: don't put code with side effects into g_assert()
7556 The g_assert() might get compiled out with the right
7557 compiler/preprocessor flags.
7559 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7561 * examples/.gitignore:
7562 examples: add cgroup test binary to .gitignore
7564 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7566 * examples/test-cgroups.c:
7567 examples: fix cgroup test build
7568 Fixes build failure caused by compiler warning:
7569 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7571 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7574 .gitignore: ignore temp files created in the course of 'make check'
7576 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7578 * gst/rtsp-server/rtsp-media.c:
7579 rtsp-media: don't loose frames handling new PLAY request
7580 If client supplied a range check if the range specifies the start point.
7581 If not, then do an accurate seek to the current position. If a start
7582 point was specified do do a key unit seek to make sure the streaming
7583 starts with decodeable frames.
7584 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7586 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7588 * gst/rtsp-server/rtsp-media.c:
7589 Revert "media: only flush when setting a new start position"
7590 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7591 We need to do the flush in all cases, demuxer block currently for
7594 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7596 * gst/rtsp-server/rtsp-media.c:
7597 media: only flush when setting a new start position
7598 Only flush the pipeline when we change the start position with
7600 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7602 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7604 * gst/rtsp-server/rtsp-stream.c:
7605 stream: set ttl-mc before adding the socket
7606 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7607 never be set on socket.
7608 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7610 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7612 * gst/rtsp-server/rtsp-media.c:
7613 media: stop thread if media is already prepared
7614 in gst_rtsp_media_prepare() the thread is not used if media is already
7615 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7617 https://bugzilla.gnome.org/show_bug.cgi?id=724182
7619 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
7622 build: Ship gst-rtsp-server.doap file
7624 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
7626 * tests/check/gst/rtspserver.c:
7627 tests: Fix another compiler warning with gcc
7629 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
7631 * gst/rtsp-server/rtsp-client.c:
7632 * gst/rtsp-server/rtsp-mount-points.c:
7633 * gst/rtsp-server/rtsp-stream.c:
7634 * tests/check/gst/client.c:
7635 rtsp-server: Fix lots of compiler warnings with clang
7637 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
7640 * gst-rtsp-server.doap:
7641 * tests/Makefile.am:
7642 configure: Synchronise with the configure scripts of the other modules
7644 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
7647 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
7649 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
7651 * gst/rtsp-server/rtsp-media.c:
7652 * gst/rtsp-server/rtsp-stream.c:
7653 Revert "rtsp-server: support build against last stable release"
7654 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
7655 Let us require 1.2.3 now, which is going to be released in a few
7658 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
7660 * gst/rtsp-server/rtsp-session-media.c:
7661 * gst/rtsp-server/rtsp-stream-transport.c:
7662 session: improve RTP-Info
7663 Ignore streams that can't generate RTP-Info instead of failing.
7664 Don't return the empty string when all streams are unconfigured but
7665 return NULL so that we don't generate and empty RTP-Info header.
7666 Improve docs a little.
7668 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
7670 * gst/rtsp-server/rtsp-session-media.c:
7671 Don't free rtpinfo GString when it is NULL
7672 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7674 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
7676 * gst/rtsp-server/rtsp-media.c:
7677 media: only set keyframe flag when modifying start
7678 Only set the keyframe flag when we modify the start position. The
7679 keyframe flag should probably be ignored when no change is requested but
7680 until we can claim this is all documented properly and all demuxer
7681 implement this, avoid setting the flag.
7682 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
7684 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
7686 * gst/rtsp-server/rtsp-thread-pool.c:
7687 thread-pool: Unref source after mainloop has quit to avoid races in GLib
7688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
7690 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
7692 * gst/rtsp-server/rtsp-stream.c:
7693 stream: handle NULL seqnum and rtptime arguments
7695 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
7697 * gst/rtsp-server/rtsp-thread-pool.c:
7698 * tests/check/gst/threadpool.c:
7699 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
7700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
7702 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
7704 * gst/rtsp-server/rtsp-stream.c:
7705 stream: add fallback for missing stats property
7706 Use a fallback when the payloader does not have a stats property
7707 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7709 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
7712 Automatic update of common submodule
7713 From f7bc1c3 to 1a07da9
7715 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
7717 * gst/rtsp-server/rtsp-stream.c:
7718 stream: don't leak stats structure
7719 Don't leak the stats structure and deal with NULL stats.
7721 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
7723 * gst/rtsp-server/rtsp-stream.c:
7724 stream: Get rtpinfo properties atomically from payloader
7725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
7727 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
7729 * gst/rtsp-server/rtsp-media.c:
7730 media: refactor state change functions and signals
7731 Make functions to set the target state and the pipeline state and emit
7732 the signals from those functions.
7734 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
7736 * gst/rtsp-server/rtsp-media.c:
7737 * gst/rtsp-server/rtsp-media.h:
7738 media: add signal to notify of pending state changes
7740 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
7742 * gst/rtsp-server/rtsp-media.c:
7743 * gst/rtsp-server/rtsp-stream.c:
7744 rtsp-server: support build against last stable release
7745 Until 1.2.3 is out with the new get_type function and we
7748 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
7750 * gst/rtsp-server/rtsp-stream.c:
7751 stream: fix compilation
7753 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
7755 * gst/rtsp-server/rtsp-media.c:
7756 * gst/rtsp-server/rtsp-media.h:
7757 * gst/rtsp-server/rtsp-stream.c:
7758 * gst/rtsp-server/rtsp-stream.h:
7759 stream: add property to configure profiles
7761 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
7763 * gst/rtsp-server/rtsp-client.c:
7764 client: let stream check supported transport
7765 Delegate the check if a transport is allowed to the stream.
7766 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
7768 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
7770 * gst/rtsp-server/rtsp-stream.c:
7771 * gst/rtsp-server/rtsp-stream.h:
7772 stream: add method to check supported transport
7773 Add a method to check if a transport is supported
7775 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
7778 configure.ac: Only check for gstreamer-check, not check
7779 We include check in gstreamer-check since quite some time now.
7781 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
7783 * gst/rtsp-server/rtsp-session-media.c:
7784 * gst/rtsp-server/rtsp-stream-transport.c:
7785 * gst/rtsp-server/rtsp-stream.c:
7786 * gst/rtsp-server/rtsp-stream.h:
7787 stream: return clock-rate from get_rtpinfo
7788 And use it to correct the rtptime to the requested start-time.
7789 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
7791 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
7793 * gst/rtsp-server/rtsp-session-media.c:
7794 * gst/rtsp-server/rtsp-stream-transport.c:
7795 * gst/rtsp-server/rtsp-stream-transport.h:
7796 session-media: calculate start-time
7798 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
7800 * gst/rtsp-server/rtsp-stream-transport.c:
7801 * gst/rtsp-server/rtsp-stream.c:
7802 * gst/rtsp-server/rtsp-stream.h:
7803 stream: also return the running-time
7804 Return the running-time in the rtpinfo as well.
7806 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
7808 * gst/rtsp-server/rtsp-client.c:
7809 * gst/rtsp-server/rtsp-session-media.c:
7810 * gst/rtsp-server/rtsp-session-media.h:
7811 * gst/rtsp-server/rtsp-stream-transport.c:
7812 * gst/rtsp-server/rtsp-stream-transport.h:
7813 session-media: let the session-media make the RTPInfo
7814 Add method to create the RTPInfo for a stream-transport.
7815 Add method to create the RTPInfo for all stream-transports in a
7817 Use the session-media RTPInfo code in client. This allows us to refactor
7818 another method to link the TCP callbacks.
7820 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7822 mount-points: sort sequence before g_sequence_lookup
7823 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
7824 sort sequence if dirty, otherwise lookup will fail.
7825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
7827 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
7830 configure: rename package from gst-rtsp to gst-rtsp-server
7831 To match git module name and avoid confusion with the
7832 rtsp lib in gst-plugins-base and rtsp plugin in -good.
7834 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
7837 configure: bump core/base/good requirement to 1.2.0
7838 Bump to released stable version and make implicit
7839 requirements explicit.
7841 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
7846 Fix broken gettext setup which is not used anyway
7848 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
7851 Automatic update of common submodule
7852 From dbedaa0 to d48bed3
7854 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
7856 * gst/rtsp-server/rtsp-client.c:
7857 * gst/rtsp-server/rtsp-media.c:
7858 * gst/rtsp-server/rtsp-media.h:
7859 media: add setup_sdp vmethod
7860 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
7861 gst_rtsp_media_setup_sdp.
7862 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
7864 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
7866 * gst/rtsp-server/rtsp-stream.c:
7867 rtsp-stream: Check return value of sscanf
7868 streamid is only valid if sscanf matched something.
7870 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
7872 * gst/rtsp-server/rtsp-client.c:
7873 rtsp-client: Fix iteration
7874 Wouldn't even enter the code block otherwise (i++ was used as the check
7875 and not the postfix).
7877 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
7879 * gst/rtsp-server/rtsp-client.c:
7880 * gst/rtsp-server/rtsp-client.h:
7881 client: add vmethod to configure media and streams
7882 Implement a vmethod that can be used to configure the media and the
7883 streams based on the current context. Handle the blocksize handling in
7884 the default handler.
7885 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
7887 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7890 Make git ignore more unit test binaries
7892 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7894 * gst/rtsp-server/rtsp-address-pool.h:
7895 * gst/rtsp-server/rtsp-auth.h:
7896 * gst/rtsp-server/rtsp-client.h:
7897 * gst/rtsp-server/rtsp-context.h:
7898 * gst/rtsp-server/rtsp-media-factory-uri.h:
7899 * gst/rtsp-server/rtsp-media-factory.h:
7900 * gst/rtsp-server/rtsp-media.h:
7901 * gst/rtsp-server/rtsp-mount-points.h:
7902 * gst/rtsp-server/rtsp-server.h:
7903 * gst/rtsp-server/rtsp-session-media.h:
7904 * gst/rtsp-server/rtsp-session-pool.h:
7905 * gst/rtsp-server/rtsp-session.h:
7906 * gst/rtsp-server/rtsp-stream-transport.h:
7907 * gst/rtsp-server/rtsp-stream.h:
7908 * gst/rtsp-server/rtsp-thread-pool.h:
7909 * gst/rtsp-server/rtsp-token.h:
7910 rtsp-server: add padding to many public structures
7911 Not mini objects though, since they are not subclassable
7912 anyway, nor kept on the stack or inlined in a structure.
7914 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7916 media: add new create_rtpbin vmethod
7917 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
7918 https://bugzilla.gnome.org/show_bug.cgi?id=719734
7920 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
7922 * tests/check/gst/media.c:
7923 tests: fix memory leak, free test's thread pool
7924 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
7926 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
7928 * gst/rtsp-server/rtsp-stream-transport.c:
7929 stream-transport: free url in finalize
7931 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
7933 * gst/rtsp-server/rtsp-media.c:
7934 media: also do state change in suspended state
7936 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
7938 * gst/rtsp-server/rtsp-client.c:
7939 * gst/rtsp-server/rtsp-media.c:
7940 media: also handle prepare and range in suspended state
7941 When we are suspended, we are already prepared.
7942 We can get the range in the suspended state.
7944 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
7946 * tests/check/Makefile.am:
7947 * tests/check/gst/sessionmedia.c:
7948 check: add test for uri in setup
7949 Added unit tests for the new functionality in GstRTSPStreamTransport.
7950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7952 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
7954 * gst/rtsp-server/rtsp-client.c:
7955 client: store setup uri and use in PLAY response
7956 Store the uri used when doing the setup and use that in the PLAY
7958 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7960 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
7962 * gst/rtsp-server/rtsp-stream-transport.c:
7963 * gst/rtsp-server/rtsp-stream-transport.h:
7964 stream-transport: add method to get/set url
7966 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
7968 * gst/rtsp-server/rtsp-client.c:
7969 client: suspend after SDP and unsuspend before PLAYING
7970 Based on patches by Ognyan Tonchev <ognyan@axis.com>
7971 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
7973 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
7975 * gst/rtsp-server/rtsp-media-factory.c:
7976 * gst/rtsp-server/rtsp-media-factory.h:
7977 * gst/rtsp-server/rtsp-media.c:
7978 * gst/rtsp-server/rtsp-media.h:
7979 * gst/rtsp-server/rtsp-session-media.c:
7980 * gst/rtsp-server/rtsp-session.c:
7981 * tests/check/gst/media.c:
7982 * tests/check/gst/mediafactory.c:
7983 media: add suspend modes
7984 Add support for different suspend modes. The stream is suspended right after
7985 producing the SDP and after PAUSE. Different suspend modes are available that
7986 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
7987 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
7988 state and RESET will bring the pipeline to the NULL state.
7989 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
7990 this means that the pipeline needs to be prerolled again.
7991 Base on patches by Ognyan Tonchev <ognyan@axis.com>
7992 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
7994 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
7996 * gst/rtsp-server/rtsp-media.c:
7997 media: start live streams in blocked state
7998 Start live streams in the blocked state and make them preroll using the
7999 messages. This ensure that no data is played by the sink until we explicitly
8000 unblock the stream right before going to PLAYING.
8001 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8003 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8005 * gst/rtsp-server/rtsp-media.c:
8006 media: refactor starting and waiting for preroll
8007 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8008 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8010 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8012 * gst/rtsp-server/rtsp-stream.c:
8013 * gst/rtsp-server/rtsp-stream.h:
8014 stream: add API to block streams
8015 Add an API to block on the streams and make it post a message.
8016 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8017 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8019 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8021 * docs/libs/Makefile.am:
8022 docs: Specify the override file
8023 Even if it's empty (for now) it avoids make distcheck complaining
8025 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8027 * gst/rtsp-server/rtsp-media.c:
8028 media: move default implementations to where they are used
8030 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8032 * gst/rtsp-server/rtsp-media.c:
8033 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8034 We need to take the state_lock when calling this method.
8036 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8038 * gst/rtsp-server/rtsp-media.c:
8039 media: handle add-added on non-bins too
8040 Handle dynamic payloaders that are not bins, as used in the unit-test.
8042 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8044 * gst/rtsp-server/rtsp-media-factory.c:
8045 * gst/rtsp-server/rtsp-media-factory.h:
8046 * gst/rtsp-server/rtsp-media.c:
8047 rtsp-media/-factory: Fix request pad name comments
8048 These must be escaped for gtk-doc to parse the comments without warnings.
8050 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8052 rtsp-media: remove transports if media is in error status
8053 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8054 trying to change to GST_STATE_NULL and media is in error status, we
8055 remove all transports.
8056 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8058 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8060 * gst/rtsp-server/rtsp-media.c:
8061 rtsp-media: use element metadata to find payloader
8062 Use the element metadata to find the payloader instead of checking
8064 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8066 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8068 rtsp-stream: add getter for payload type
8069 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8070 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8071 element and create the stream with this one instead of the dynpay%d
8073 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8075 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8077 * gst/rtsp-server/rtsp-client.c:
8078 * gst/rtsp-server/rtsp-context.h:
8079 * gst/rtsp-server/rtsp-media.c:
8080 * gst/rtsp-server/rtsp-mount-points.c:
8081 * gst/rtsp-server/rtsp-server.c:
8082 * gst/rtsp-server/rtsp-token.c:
8083 rtsp-*: Refer to NULL as a constant in comments
8085 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8087 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8089 rtsp-*: Fix type name typos in comments
8090 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8091 * rtsp-auth: Refer to part of constant name as text
8092 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8093 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8094 * rtsp-stream: Fix typo when refering to GstBin
8095 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8097 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8100 * docs/libs/gst-rtsp-server-docs.sgml:
8101 * docs/libs/gst-rtsp-server-sections.txt:
8102 docs: Improve documentation
8103 * Include annotation-glossary to quiet gtk-doc
8104 * Rename remaining ClientState -> Context
8105 * Rename object hierarchy file
8106 * Remove stale chapter references
8107 * Add missing function and object references
8108 * Include missing GstRTSPAddressPoolResult
8109 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8111 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8113 * gst/rtsp-server/rtsp-client.c:
8114 * gst/rtsp-server/rtsp-server.c:
8115 * gst/rtsp-server/rtsp-session-pool.c:
8116 * gst/rtsp-server/rtsp-session.c:
8117 * gst/rtsp-server/rtsp-stream.c:
8118 rtsp-server: sprinkle some allow-none annotations for g-i
8120 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8122 * gst/rtsp-server/rtsp-stream.c:
8123 * gst/rtsp-server/rtsp-stream.h:
8124 stream: add method to filter transports
8125 Add a method to safely iterate and collect the stream transports
8126 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8128 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8130 * gst/rtsp-server/rtsp-client.c:
8131 * gst/rtsp-server/rtsp-server.c:
8132 * gst/rtsp-server/rtsp-session-pool.c:
8133 * gst/rtsp-server/rtsp-session.c:
8134 rtsp: allow NULL func in filters
8135 Passing a null function make the filters return a list of
8138 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8140 * gst/rtsp-server/rtsp-address-pool.c:
8141 * tests/check/gst/addresspool.c:
8142 address-pool: fix address increment
8143 Use a guint instead of guint8 to increment the address. It's still not
8144 completely correct because a guint might not be able to hold the complete
8145 address range, but that's an enhacement for later.
8146 Add unit test to test improved behaviour.
8147 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8149 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8151 * gst/rtsp-server/rtsp-client.c:
8152 * tests/check/gst/client.c:
8153 client: allow absolute path in requests
8154 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8156 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8158 * gst/rtsp-server/rtsp-client.c:
8159 * gst/rtsp-server/rtsp-client.h:
8160 client: make make_path_from_uri a vmethod
8162 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8164 * docs/libs/gst-rtsp-server-sections.txt:
8165 * gst/rtsp-server/rtsp-stream.c:
8166 * gst/rtsp-server/rtsp-stream.h:
8167 * tests/check/Makefile.am:
8168 * tests/check/gst/stream.c:
8169 stream: Add functions to get rtp and rtcp sockets
8170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8172 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8174 * gst/rtsp-server/rtsp-context.c:
8175 * gst/rtsp-server/rtsp-context.h:
8176 context: defing a GType for the context
8177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8179 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8181 * gst/rtsp-server/Makefile.am:
8182 * gst/rtsp-server/rtsp-auth.c:
8183 * gst/rtsp-server/rtsp-context.c:
8184 * gst/rtsp-server/rtsp-media.c:
8185 * gst/rtsp-server/rtsp-mount-points.c:
8186 * gst/rtsp-server/rtsp-server.h:
8187 * gst/rtsp-server/rtsp-session-media.c:
8188 * gst/rtsp-server/rtsp-session.c:
8189 * gst/rtsp-server/rtsp-stream.c:
8190 Fixed several GIR warnings
8192 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8194 * gst/rtsp-server/rtsp-auth.c:
8197 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8199 * tests/check/Makefile.am:
8200 * tests/check/gst/token.c:
8201 tests: Add unit tests for token
8202 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8204 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8206 * gst/rtsp-server/rtsp-token.c:
8207 token: Validate args for gst_rtsp_token_is_allowed
8208 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8210 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8212 * gst/rtsp-server/rtsp-token.c:
8213 token: Fix bug when creating empty token
8214 We always want to have a valid GstStructure in the token.
8215 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8217 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8219 * gst/rtsp-server/rtsp-thread-pool.c:
8220 thread-pool: avoid race in shutdown
8221 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8222 don't actually stop the mainloop ever. Solve this race by adding an idle source
8223 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8224 if quit was called before we started it.
8226 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8228 * tests/check/Makefile.am:
8229 * tests/check/gst/permissions.c:
8230 tests: Add unit tests for permissions
8231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8233 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8235 * tests/check/gst/mediafactory.c:
8236 tests: Test mediafactory permissions
8237 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8239 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8241 * gst/rtsp-server/rtsp-permissions.c:
8242 permissions: Fix refcounting when adding/removing roles
8243 Previously a role that was removed was unreffed twice, and when
8244 replacing an existing role the replaced role was freed while still being
8245 referenced. Both bugs are now fixed.
8246 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8248 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8250 * tests/check/gst/media.c:
8251 * tests/check/gst/mediafactory.c:
8252 * tests/check/gst/rtspserver.c:
8253 tests: Check gst_rtsp_url_parse return value
8254 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8256 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8259 Automatic update of common submodule
8260 From 865aa20 to dbedaa0
8262 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8264 * gst/rtsp-server/rtsp-server.c:
8265 rtsp-server: Fix socket leak
8266 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8268 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8270 * gst/rtsp-server/rtsp-session-pool.c:
8271 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8272 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8274 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8276 * examples/.gitignore:
8277 * examples/test-video.c:
8278 examples: fix compilation when WITH_AUTH is defined
8279 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8281 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8284 gitignore: Add new test binary
8286 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8288 * tests/check/Makefile.am:
8289 * tests/check/gst/threadpool.c:
8290 thread-pool: Add unit test for the thread pools
8291 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8293 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8295 * gst/rtsp-server/rtsp-thread-pool.c:
8296 thread-pool: Fix thread leak when reusing threads
8297 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8299 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8301 * gst/rtsp-server/rtsp-server.c:
8302 * tests/check/gst/rtspserver.c:
8303 tests: fixed racy behavior in rtspserver tests
8304 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8306 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8308 * tests/check/gst/addresspool.c:
8309 tests: Improve address pool unit tests
8310 Add a range with mixed IPV4 and IPV6 addresses to pool.
8311 Get an IPV4 address from an IPV6-only pool.
8312 Get an IPV6 address from an IPV4-only pool.
8313 Reserve a IPV6 address from an IPV4-only pool.
8314 Check for unicast addresses in multicast-only pool.
8315 Check for unicast addresses in uni-/multicast-mixed pool.
8316 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8318 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8320 * gst/rtsp-server/rtsp-client.c:
8321 client: append query string in PAUSE/PLAY/TEARDOWN as well
8323 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8325 * gst/rtsp-server/rtsp-client.c:
8326 client: Add query to control path
8327 If the SETUP url contains a query it must be appended to the control
8328 path so that it matches any already created stream in the media. The
8329 query will also be appended to the session media path.
8331 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8333 * gst/rtsp-server/rtsp-media.c:
8334 rtsp-media: remove old line
8336 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8338 * gst/rtsp-server/rtsp-stream.c:
8339 stream: Correct control comparison
8340 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8342 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8344 * gst/rtsp-server/rtsp-media.c:
8345 media: Check dynamically if the pipeline supports seeking
8346 We should not depend on whether or not the pipeline state change
8347 returned NO_PREROLL or not. A media could dynamically change its
8348 element and switch from seekable to non seekable so it's best to test
8349 the seekable nature of the pipeline dynamically when we try to do a seek.
8351 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8353 * gst/rtsp-server/rtsp-media.c:
8354 media: Return FALSE if seeking is not supported
8356 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8358 * gst/rtsp-server/rtsp-media.c:
8359 rtsp-media: don't seek accurate by default
8360 Accurate seeking is perhaps a little overkill in the most common situation and
8361 causes some formats (mp3) over slow media to seek extremely slowly.
8363 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8365 * tests/check/gst/rtspserver.c:
8366 tests: fix unit test
8367 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8369 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8371 * gst/rtsp-server/rtsp-client.c:
8372 client: Reply 400 if media cannot be constructed
8373 Reply 400 Bad Request instead of 503 Service Unavailable if media
8374 cannot be constructed in SETUP.
8375 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8377 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8379 * gst/rtsp-server/rtsp-client.c:
8380 client: Send setup reply once only
8381 If find_media() failed in handle_setup_request() two replies was sent.
8382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8384 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8387 Automatic update of common submodule
8388 From 6b03ba7 to 865aa20
8390 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8392 * gst/rtsp-server/rtsp-server.c:
8393 server: Emit client-connected signal earlier
8394 Emit client-connected before the client ref is given to a GSource,
8395 otherwise client-connected can be emitted after the client object has
8398 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8400 * gst/rtsp-server/rtsp-address-pool.c:
8401 * gst/rtsp-server/rtsp-address-pool.h:
8402 * gst/rtsp-server/rtsp-stream.c:
8403 * tests/check/gst/addresspool.c:
8404 addresspool: return reason of failure
8405 Let gst_rtsp_address_pool_reserve_address() return the reason why
8406 the address could not be reserved.
8407 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8409 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8412 autogen.sh: Sync behaviour with other GStreamer modules
8413 Allows building from outside of tree amongst other things
8415 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8418 Automatic update of common submodule
8419 From b613661 to 6b03ba7
8421 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8424 Automatic update of common submodule
8425 From 74a6857 to b613661
8427 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8430 Automatic update of common submodule
8431 From 01a7a46 to 74a6857
8433 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8435 * gst/rtsp-server/rtsp-client.c:
8436 client: Do not read beyond end of path string
8437 If the setup was done without a control url, make sure we don't try to read the
8438 non-existing control string and crash.
8440 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-client.c:
8443 client: Fix RTPInfo header
8444 Refactor the method to make the content_base.
8445 Use the content-base and the control url to construct the RTPInfo
8448 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8450 * gst/rtsp-server/rtsp-client.c:
8451 client: map url to path only in describe
8452 Only map the request url to a path in the DESCRIBE method. The SDP then
8453 contains the base and control urls that should be used to SETUP/PAUSE/
8454 PLAY/TEARDOWN the media.
8456 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8458 * gst/rtsp-server/rtsp-client.c:
8459 Revert "client: map URL to path in requests"
8460 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8461 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8462 contains the base and control urls which are used in the SETUP, PLAY,
8463 PAUSE and TEARDOWN requests.
8465 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-client.c:
8468 client: map URL to path in requests
8470 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8472 * gst/rtsp-server/rtsp-client.c:
8473 * gst/rtsp-server/rtsp-mount-points.c:
8474 * gst/rtsp-server/rtsp-mount-points.h:
8475 mount-points: make vmethod to make path from uri
8476 Make a vmethod to transform an url into a path. The path is then used to lookup
8477 the factory. This makes it possible to also use other bits of the url, such as
8478 the query parameters, to locate the factory.
8480 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8482 * gst/rtsp-server/rtsp-thread-pool.c:
8483 * gst/rtsp-server/rtsp-thread-pool.h:
8484 thread-pool: Add cleanup to wait for the threadpool to finish
8485 Also fix race condition if two threads are asking for the first
8486 thread from the thread pool at once. This would case two internal
8487 GThreadPools to be created.
8488 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8490 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8492 * gst/rtsp-server/rtsp-client.c:
8493 * tests/check/gst/client.c:
8494 client: free threadpool
8495 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8497 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8499 * tests/check/gst/mountpoints.c:
8500 mountpoints tests: unref matched factories
8501 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8503 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8505 * tests/check/gst/media.c:
8506 media tests: unref thread pool and caps
8507 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8509 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8511 * gst/rtsp-server/rtsp-auth.c:
8512 * gst/rtsp-server/rtsp-media-factory.c:
8513 * gst/rtsp-server/rtsp-media.c:
8514 auth, media, media-factory: unref permissions
8515 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8517 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8519 * examples/Makefile.am:
8520 Makefile: add rule for appsrc example
8522 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8524 * examples/test-appsrc.c:
8525 tests: add appsrc example
8526 Add an example on how to use appsrc to feed the server pipeline with data.
8528 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8530 * gst/rtsp-server/rtsp-client.c:
8531 rtsp-client: remove query part from content-base string
8532 Make sure that after the control url has been resolved, it's
8533 not a part of the query-string.
8534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8536 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8538 * gst/rtsp-server/rtsp-client.c:
8539 client: don't check url in response
8540 There is no url or method in the response to check
8542 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8544 * gst/rtsp-server/rtsp-client.c:
8545 * gst/rtsp-server/rtsp-client.h:
8546 Add handle-response signal for when we receive a GET_PARAMETER response
8548 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8550 * gst/rtsp-server/rtsp-server.c:
8551 Fix gst_rtsp_server_client_filter, using wrong variable type
8553 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8555 * gst/rtsp-server/rtsp-media-factory-uri.c:
8556 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8557 For AAC we need to check for framed=true instead of parsed=true.
8558 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8560 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8562 * gst/rtsp-server/rtsp-stream.c:
8563 stream: optimize pipeline for protocols
8564 When TCP is not an allowed protocol for the stream, avoid creating the
8565 appsrc/appsink/queue and tee elements.
8567 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * gst/rtsp-server/rtsp-media.c:
8570 media: set protocols on streams
8572 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8574 * gst/rtsp-server/rtsp-client.c:
8575 client: use protocols supported by stream
8577 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8579 * gst/rtsp-server/rtsp-media-factory.c:
8580 * gst/rtsp-server/rtsp-media.c:
8581 * gst/rtsp-server/rtsp-stream.c:
8582 media-factory: allow all protocols
8584 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/rtsp-media.c:
8587 media: configure protocols in new streams
8589 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * gst/rtsp-server/rtsp-stream.c:
8592 * gst/rtsp-server/rtsp-stream.h:
8593 stream: add protocols property
8595 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8597 * gst/rtsp-server/rtsp-media.c:
8598 rtsp-media: send state in "new-state" signal
8599 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8601 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8604 build: add subdir-objects to AM_INIT_AUTOMAKE
8605 Fixes warnings with automake 1.14
8606 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8608 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8610 * docs/libs/gst-rtsp-server-sections.txt:
8611 * gst/rtsp-server/rtsp-client.c:
8612 * gst/rtsp-server/rtsp-server.c:
8613 * gst/rtsp-server/rtsp-server.h:
8614 server: add method to iterate clients of server
8616 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8618 * gst/rtsp-server/rtsp-media.c:
8619 * gst/rtsp-server/rtsp-media.h:
8620 Add vmethod for rtsp-media subclass to access rtpbin
8622 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8624 * gst/rtsp-server/rtsp-client.h:
8625 small documentation fix
8627 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8629 * gst/rtsp-server/rtsp-client.c:
8630 Do not take range header if range is invalid
8632 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8634 * docs/libs/gst-rtsp-server-sections.txt:
8635 * gst/rtsp-server/rtsp-media.c:
8636 media: add docs for new method
8638 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-media.c:
8641 * gst/rtsp-server/rtsp-media.h:
8642 Add API to rtsp-media set the pipeline's state
8644 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8646 * gst/rtsp-server/rtsp-media.c:
8647 Update current position/duration when gst_rtsp_media_get_range_string is called
8649 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8651 * examples/test-cgroups.c:
8652 tests: add some more docs
8654 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8656 * examples/test-cgroups.c:
8657 * gst/rtsp-server/Makefile.am:
8658 * gst/rtsp-server/rtsp-auth.c:
8659 * gst/rtsp-server/rtsp-auth.h:
8660 * gst/rtsp-server/rtsp-client.c:
8661 * gst/rtsp-server/rtsp-client.h:
8662 * gst/rtsp-server/rtsp-context.c:
8663 * gst/rtsp-server/rtsp-context.h:
8664 * gst/rtsp-server/rtsp-params.c:
8665 * gst/rtsp-server/rtsp-params.h:
8666 * gst/rtsp-server/rtsp-server.c:
8667 * gst/rtsp-server/rtsp-thread-pool.c:
8668 * gst/rtsp-server/rtsp-thread-pool.h:
8669 * tests/check/gst/client.c:
8670 ClientState -> Context
8671 Rename the clientstate to context and put the code in a separate file.
8673 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * examples/test-auth.c:
8676 * gst/rtsp-server/rtsp-auth.c:
8677 * gst/rtsp-server/rtsp-auth.h:
8678 auth: add support for default token
8679 The default token is used when the user is not authenticated and can be used to
8680 give minimal permissions.
8682 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8684 * examples/test-auth.c:
8685 * gst/rtsp-server/rtsp-auth.c:
8686 auth: use defines when possible
8688 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8690 * gst/rtsp-server/rtsp-address-pool.c:
8691 address-pool: improve docs
8693 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8695 * gst/rtsp-server/rtsp-permissions.c:
8696 permissions: add the role to the copy
8698 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
8700 * gst/rtsp-server/rtsp-permissions.c:
8701 permissions: Also copy the roles
8703 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
8705 * gst/rtsp-server/rtsp-permissions.c:
8706 permissions: Make it build
8708 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8710 * gst/rtsp-server/rtsp-address-pool.h:
8713 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8715 * docs/libs/gst-rtsp-server-sections.txt:
8716 * gst/rtsp-server/rtsp-auth.c:
8717 * gst/rtsp-server/rtsp-auth.h:
8718 * gst/rtsp-server/rtsp-media.c:
8719 * gst/rtsp-server/rtsp-session-media.c:
8720 * gst/rtsp-server/rtsp-stream-transport.c:
8721 * gst/rtsp-server/rtsp-stream-transport.h:
8722 * gst/rtsp-server/rtsp-stream.c:
8723 * tests/check/gst/client.c:
8726 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8728 * docs/libs/gst-rtsp-server-sections.txt:
8729 * gst/rtsp-server/rtsp-address-pool.c:
8730 * gst/rtsp-server/rtsp-address-pool.h:
8731 * tests/check/gst/addresspool.c:
8732 * tests/check/gst/rtspserver.c:
8733 address-pool: cleanups
8734 Remove redundant method, improve docs.
8736 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8738 * docs/libs/gst-rtsp-server-sections.txt:
8739 * gst/rtsp-server/rtsp-auth.h:
8740 * gst/rtsp-server/rtsp-permissions.c:
8741 * gst/rtsp-server/rtsp-permissions.h:
8742 * gst/rtsp-server/rtsp-token.c:
8745 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8747 * gst/rtsp-server/rtsp-permissions.c:
8748 permissions: implement _remove_role
8750 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8752 * gst/rtsp-server/rtsp-permissions.c:
8753 permissions: update docs
8755 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8757 * tests/check/gst/client.c:
8758 tests: simplify tests
8759 Client settings are now disabled by default so we don't need an auth
8760 module to disable them.
8762 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8764 * gst/rtsp-server/rtsp-auth.c:
8765 auth: add default authorizations
8766 When no auth module is specified, use our table of defaults to look up the
8767 default value of the check instead of always allowing everything. This was
8768 we can disallow client settings by default.
8770 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8773 README: update readme
8775 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8777 * gst/rtsp-server/rtsp-thread-pool.c:
8778 * gst/rtsp-server/rtsp-thread-pool.h:
8779 thread-pool: add more docs
8781 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8783 * gst/rtsp-server/rtsp-thread-pool.c:
8784 * gst/rtsp-server/rtsp-thread-pool.h:
8785 thread-pool: fix race in thread reuse
8786 If we try to reuse a thread right after we made it stop, we end up using a
8787 stopped thread. Catch this case and only reuse threads that are not stopping.
8789 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8791 * gst/rtsp-server/rtsp-server.c:
8792 server: add small debug
8794 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8796 * tests/check/gst/client.c:
8798 Add some permissions to media so we can use the auth and enable
8801 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8803 * gst/rtsp-server/rtsp-client.c:
8804 client: support pushed context in handle_request
8805 If we already have a pushed state, reuse it and add our own things. This makes
8806 it easier to write tests.
8808 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8810 * gst/rtsp-server/rtsp-auth.c:
8811 auth: don't auth on methods
8812 Don't authorize on methods anymore but on the resources that we
8813 try to access, this is more flexible.
8814 Move the authorization checks to where they are needed and let the
8815 check return the response on error.
8817 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8819 * gst/rtsp-server/rtsp-mount-points.c:
8820 mount-points: add some debug
8822 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8824 * tests/check/gst/client.c:
8825 tests: almost fix test
8827 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8829 * gst/rtsp-server/rtsp-auth.c:
8830 * gst/rtsp-server/rtsp-auth.h:
8831 * gst/rtsp-server/rtsp-client.c:
8832 * gst/rtsp-server/rtsp-client.h:
8833 * gst/rtsp-server/rtsp-server.c:
8834 * gst/rtsp-server/rtsp-server.h:
8835 auth: let the auth module check client_settings
8836 Let the auth module decide if client settings are allowed for the
8839 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8841 * gst/rtsp-server/rtsp-token.c:
8842 * gst/rtsp-server/rtsp-token.h:
8843 token: add method to check boolean permission
8845 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8847 * examples/test-auth.c:
8848 * examples/test-cgroups.c:
8849 * gst/rtsp-server/rtsp-token.c:
8850 * gst/rtsp-server/rtsp-token.h:
8851 token: simplify token constructor
8852 Use variable arguments to make easier API.
8854 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8856 * examples/test-auth.c:
8857 * examples/test-cgroups.c:
8858 * gst/rtsp-server/rtsp-media-factory.c:
8859 * gst/rtsp-server/rtsp-media-factory.h:
8860 media-factory: add convenience API for factory
8862 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8864 * examples/test-auth.c:
8865 * examples/test-cgroups.c:
8866 * gst/rtsp-server/rtsp-permissions.c:
8867 * gst/rtsp-server/rtsp-permissions.h:
8868 permissions: simplify API a little
8869 Avoid passing GstStructure in the add_role method, use varargs instead
8870 to construct the structure behind the scenes. We can then also use the
8871 structure name as the role and simplify some more logic.
8873 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8875 * gst/rtsp-server/rtsp-auth.c:
8878 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/rtsp-auth.c:
8881 * gst/rtsp-server/rtsp-auth.h:
8882 * gst/rtsp-server/rtsp-client.c:
8883 auth: handle unauthorized response
8884 Move handling of the unauthorized response to the auth module, it can add
8885 the appropriate headers to request authorization for the required method
8886 much better than the client.
8888 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8890 * gst/rtsp-server/rtsp-client.c:
8891 * gst/rtsp-server/rtsp-client.h:
8892 client: allow for sending any message, not only requests
8893 Change the _send_request() method to _send_message() so that we
8894 can both send requests and replies.
8896 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8898 * docs/libs/gst-rtsp-server-sections.txt:
8899 * gst/rtsp-server/rtsp-server.h:
8902 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8904 * examples/test-video.c:
8905 * gst/rtsp-server/rtsp-auth.c:
8906 * gst/rtsp-server/rtsp-auth.h:
8907 * gst/rtsp-server/rtsp-server.c:
8908 * gst/rtsp-server/rtsp-server.h:
8909 auth: move TLS handling to auth module
8910 Remove the TLS settings on the server and move it to the auth module because
8911 that is where security related bits go.
8913 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8915 * gst/rtsp-server/rtsp-client.c:
8916 * gst/rtsp-server/rtsp-client.h:
8917 client: add state push/pop
8919 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8921 * gst/rtsp-server/rtsp-client.c:
8922 * gst/rtsp-server/rtsp-client.h:
8923 client: add connection to state
8925 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8927 * gst/rtsp-server/rtsp-mount-points.c:
8928 mount-points: fix debug
8930 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8932 * tests/check/gst/media.c:
8933 tests: fix media test
8935 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8937 * gst/rtsp-server/rtsp-thread-pool.c:
8938 thread-pool: we don't require a state
8940 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8942 * gst/rtsp-server/rtsp-server.c:
8943 server: let context ref the server
8944 So that we don't risk losing the server object early anc crash.
8946 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8948 * tests/check/gst/client.c:
8949 tests: fix client test
8951 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8954 * docs/libs/gst-rtsp-server-docs.sgml:
8955 * docs/libs/gst-rtsp-server-sections.txt:
8956 * gst/rtsp-server/rtsp-address-pool.c:
8957 * gst/rtsp-server/rtsp-auth.c:
8958 * gst/rtsp-server/rtsp-client.c:
8959 * gst/rtsp-server/rtsp-client.h:
8960 * gst/rtsp-server/rtsp-media-factory-uri.c:
8961 * gst/rtsp-server/rtsp-media-factory.c:
8962 * gst/rtsp-server/rtsp-media-factory.h:
8963 * gst/rtsp-server/rtsp-media.c:
8964 * gst/rtsp-server/rtsp-mount-points.c:
8965 * gst/rtsp-server/rtsp-params.c:
8966 * gst/rtsp-server/rtsp-permissions.c:
8967 * gst/rtsp-server/rtsp-sdp.c:
8968 * gst/rtsp-server/rtsp-server.c:
8969 * gst/rtsp-server/rtsp-server.h:
8970 * gst/rtsp-server/rtsp-session-media.c:
8971 * gst/rtsp-server/rtsp-session-pool.c:
8972 * gst/rtsp-server/rtsp-session.c:
8973 * gst/rtsp-server/rtsp-stream-transport.c:
8974 * gst/rtsp-server/rtsp-stream.c:
8975 * gst/rtsp-server/rtsp-thread-pool.c:
8976 * gst/rtsp-server/rtsp-token.c:
8979 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8981 * gst/rtsp-server/rtsp-session-pool.c:
8982 * gst/rtsp-server/rtsp-session-pool.h:
8983 session-pool: make vmethod to create a session
8984 Make a vmethod to create a sessions so that subclasses can create
8985 custom session objects
8987 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8989 * gst/rtsp-server/rtsp-auth.c:
8990 * gst/rtsp-server/rtsp-media-factory.h:
8991 * gst/rtsp-server/rtsp-media.h:
8992 * gst/rtsp-server/rtsp-mount-points.h:
8993 * gst/rtsp-server/rtsp-session-pool.h:
8994 * gst/rtsp-server/rtsp-stream.h:
8997 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8999 * docs/libs/gst-rtsp-server-docs.sgml:
9000 * docs/libs/gst-rtsp-server-sections.txt:
9001 * gst/rtsp-server/rtsp-address-pool.c:
9002 * gst/rtsp-server/rtsp-address-pool.h:
9003 * gst/rtsp-server/rtsp-auth.c:
9004 * gst/rtsp-server/rtsp-client.h:
9005 * gst/rtsp-server/rtsp-media-factory.h:
9006 * gst/rtsp-server/rtsp-media.c:
9007 * gst/rtsp-server/rtsp-media.h:
9008 * gst/rtsp-server/rtsp-permissions.c:
9009 * gst/rtsp-server/rtsp-permissions.h:
9010 * gst/rtsp-server/rtsp-server.h:
9011 * gst/rtsp-server/rtsp-session-media.c:
9012 * gst/rtsp-server/rtsp-session-media.h:
9013 * gst/rtsp-server/rtsp-session-pool.h:
9014 * gst/rtsp-server/rtsp-session.h:
9015 * gst/rtsp-server/rtsp-stream-transport.h:
9016 * gst/rtsp-server/rtsp-stream.c:
9017 * gst/rtsp-server/rtsp-thread-pool.h:
9020 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9023 * examples/Makefile.am:
9024 configure: compile cgroup example conditionally
9025 Only compile the cgroup example when we have libcgroup
9027 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9030 * examples/Makefile.am:
9031 * examples/test-cgroups.c:
9032 examples: add cgroups example
9034 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9036 * tests/check/gst/rtspserver.c:
9037 tests: fix compilation
9039 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9041 * gst/rtsp-server/rtsp-thread-pool.c:
9042 thread-pool: fix vmethod invocation
9044 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * gst/rtsp-server/rtsp-thread-pool.c:
9047 * gst/rtsp-server/rtsp-thread-pool.h:
9048 thread-pool: store thread type in thread
9050 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9052 * gst/rtsp-server/rtsp-client.c:
9053 client: pass thread from pool to media _prepare
9054 Get a thread from the configured threadpool and pass it to the prepare method of
9057 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9059 * gst/rtsp-server/rtsp-media.c:
9060 * gst/rtsp-server/rtsp-media.h:
9061 media: Accept a thread in _prepare
9062 Remove out own threadpool handling and use the provided thread and
9063 maincontext for the bus messages and the state changes.
9065 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9067 * gst/rtsp-server/rtsp-server.c:
9068 server: configure client thread pool
9070 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9072 * gst/rtsp-server/rtsp-client.c:
9073 * gst/rtsp-server/rtsp-client.h:
9074 client: add method to configure thread pool
9076 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9078 * gst/rtsp-server/rtsp-client.h:
9079 * gst/rtsp-server/rtsp-server.c:
9080 * gst/rtsp-server/rtsp-server.h:
9081 server: use thread pool
9082 Use the thread pool instead of doing our own thing.
9084 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9086 * gst/rtsp-server/Makefile.am:
9087 * gst/rtsp-server/rtsp-thread-pool.c:
9088 * gst/rtsp-server/rtsp-thread-pool.h:
9089 thread-pool: add object to manage threads
9090 Add an object to manage the client and media threads.
9092 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9094 * gst/rtsp-server/rtsp-auth.c:
9095 auth: debug authorization check
9097 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9099 * gst/rtsp-server/rtsp-media.c:
9100 media: start media pipeline in context
9101 Start the media pipeline in the provided context (or our default one
9102 when NULL). This makes sure that we run the bus thread in this context and that
9103 all media threads are children of this context.
9105 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-media-factory.c:
9108 factory: pass permissions to media by default
9110 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * examples/test-auth.c:
9113 test: add permissions to auth test
9114 Ass some permissions to the media factory in the test.
9116 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9118 * gst/rtsp-server/rtsp-auth.c:
9119 * gst/rtsp-server/rtsp-auth.h:
9120 * gst/rtsp-server/rtsp-client.c:
9121 auth: simplify auth checks
9122 Remove client from methods, it's now in the state
9123 Perform the check specified by the string, use the information from the
9124 thread local context.
9126 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * gst/rtsp-server/rtsp-client.c:
9129 * gst/rtsp-server/rtsp-client.h:
9130 client: add state to current thread
9131 Add the client to the ClientState object.
9132 Place the ClientState on the current thread.
9134 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9136 * gst/rtsp-server/rtsp-media-factory.c:
9137 * gst/rtsp-server/rtsp-media-factory.h:
9138 * gst/rtsp-server/rtsp-media.c:
9139 * gst/rtsp-server/rtsp-media.h:
9140 media: make it possible to set permissions
9141 Make it possible to set permissions on media and media factory objects
9143 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9145 * gst/rtsp-server/Makefile.am:
9146 * gst/rtsp-server/rtsp-permissions.c:
9147 * gst/rtsp-server/rtsp-permissions.h:
9148 permissions: add permissions object
9149 Add a mini object to store permissions based on a role.
9151 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9153 * examples/test-auth.c:
9154 * gst/rtsp-server/rtsp-auth.c:
9155 * gst/rtsp-server/rtsp-auth.h:
9156 * gst/rtsp-server/rtsp-client.c:
9157 auth: add auth checks
9158 Add an enum with auth checks and implement the checks in the auth object.
9159 Perform the checks from the client.
9161 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9163 * examples/test-auth.c:
9164 * gst/rtsp-server/rtsp-auth.c:
9165 * gst/rtsp-server/rtsp-auth.h:
9166 * gst/rtsp-server/rtsp-client.h:
9167 auth: use the token after authentication
9168 After we authenticated a user, keep the Token around in the state.
9170 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9172 * gst/rtsp-server/rtsp-client.c:
9173 * gst/rtsp-server/rtsp-media.c:
9174 * gst/rtsp-server/rtsp-media.h:
9175 * tests/check/gst/media.c:
9176 media: add optional context for bus messages
9177 Add an optional mainloop to _prepare that will handle the bus messages instead
9178 of always using the shared mainloop.
9180 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9182 * gst/rtsp-server/Makefile.am:
9183 * gst/rtsp-server/rtsp-token.c:
9184 * gst/rtsp-server/rtsp-token.h:
9185 token: add authorization token
9186 Add a simply miniobject that contains the authorizations. The object contains a
9187 GstStructure that hold all authorization fields. When a user is authenticated,
9188 the auth module will create a Token for the user. The token is then used to
9189 check what operations the user is allowed to do and various other configuration
9192 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9194 * examples/test-auth.c:
9195 * gst/rtsp-server/rtsp-auth.c:
9196 * gst/rtsp-server/rtsp-auth.h:
9197 * gst/rtsp-server/rtsp-client.c:
9198 * gst/rtsp-server/rtsp-client.h:
9199 * gst/rtsp-server/rtsp-media-factory.c:
9200 * gst/rtsp-server/rtsp-media-factory.h:
9201 * gst/rtsp-server/rtsp-media.c:
9202 * gst/rtsp-server/rtsp-media.h:
9203 auth: remove auth from media and factory
9204 Remove the auth object from media and factory. We want to have the RTSPClient
9205 authenticate and authorize resources, there is no need to place another auth
9206 manager on the media/factory.
9208 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9210 * examples/test-auth.c:
9211 * gst/rtsp-server/rtsp-auth.c:
9212 * gst/rtsp-server/rtsp-auth.h:
9213 * gst/rtsp-server/rtsp-client.h:
9214 auth: add support for multiple basic auth tokens
9215 Make it possible to add multiple basic authorisation tokens to one authorization
9216 object. Associate with each token an authorization group that will define what
9217 capabilities are allowed.
9219 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9221 * gst/rtsp-server/rtsp-client.c:
9222 client: error out on non-aggregate control
9223 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9225 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9227 * gst/rtsp-server/rtsp-client.c:
9228 client: rework setup request a little
9229 Cache the media in DESCRIBE based on the longest matching path with the uri
9230 that we can find in the mount points.
9231 Rework the setup request a little to get the media from the session or from
9232 the longest matching path, this way we can derive the control string as
9233 everything after the path instead of hardcoding it.
9234 Find the stream based on the control string and only open a session when all
9237 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9239 * gst/rtsp-server/rtsp-media.c:
9240 * gst/rtsp-server/rtsp-media.h:
9241 media: add method to find a stream by control url
9243 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9245 * gst/rtsp-server/rtsp-stream.c:
9246 * gst/rtsp-server/rtsp-stream.h:
9247 stream: add method to check control url of stream
9249 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9251 * gst/rtsp-server/rtsp-client.c:
9252 * gst/rtsp-server/rtsp-session-media.c:
9253 * gst/rtsp-server/rtsp-session-media.h:
9254 * gst/rtsp-server/rtsp-session.c:
9255 * gst/rtsp-server/rtsp-session.h:
9256 session: use path matching for session media
9257 Use a path string instead of a uri to lookup session media in the sessions. Also
9258 use path matching to find the largest possible path that matches.
9260 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9262 * gst/rtsp-server/rtsp-client.c:
9263 * gst/rtsp-server/rtsp-mount-points.c:
9264 * gst/rtsp-server/rtsp-mount-points.h:
9265 * tests/check/gst/mountpoints.c:
9266 mount-points: remove useless vmethod
9267 Making lookups in the mount points should not be done with a URL, if there is a
9268 mapping to be done from URL to mount points, we'll need to do it somewhere
9271 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9273 * gst/rtsp-server/rtsp-mount-points.c:
9274 * gst/rtsp-server/rtsp-mount-points.h:
9275 * tests/check/gst/mountpoints.c:
9276 mount-points: improve mount point searching
9277 Use a GSequence to keep track of the mount points.
9278 Match a URL to the longest matching registered mount point. This should be the
9279 URL to perform aggreagate control and the remainder is the stream specific
9281 Add some unit tests for this.
9283 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9285 * gst/rtsp-server/Makefile.am:
9286 rtsp-server: Allow building of static library
9288 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9290 * tests/check/gst/mediafactory.c:
9291 tests: fix compilation
9293 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9295 * gst/rtsp-server/rtsp-sdp.c:
9296 sdp: get control string from stream
9297 Use the control string as configured in the stream.
9299 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9301 * gst/rtsp-server/rtsp-stream.c:
9302 * gst/rtsp-server/rtsp-stream.h:
9303 stream: add methods and property to set control string
9305 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9307 * gst/rtsp-server/rtsp-client.c:
9309 Rename variables for clarity
9310 Keep media in state when we can
9312 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9314 * gst/rtsp-server/rtsp-client.c:
9315 * gst/rtsp-server/rtsp-stream.c:
9316 * gst/rtsp-server/rtsp-stream.h:
9317 stream: add more support for IPv6
9318 Rename _get_address to _get_multicast_address in GstRTSPStream to
9319 make it clear that this function only deals with multicast.
9320 Make it possible to have both an IPv4 and IPv6 multicast address on
9321 a stream. Give the client an IPv4 or IPv6 address depending on the
9322 address it used to connect to the server.
9323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9325 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9327 * gst/rtsp-server/rtsp-client.c:
9330 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9332 * gst/rtsp-server/rtsp-stream.c:
9333 stream: handle failed port allocation
9334 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9335 can't allocate any family at all. Also keep track of what port families we
9337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9339 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9341 * gst/rtsp-server/rtsp-stream.c:
9342 stream: improve docs
9344 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9346 * gst/rtsp-server/rtsp-stream-transport.c:
9347 stream-transport: remove old if 0 block
9349 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9351 * tests/check/gst/client.c:
9353 gst_rtsp_client_get_uri() has been removed
9354 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9356 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9358 * gst/rtsp-server/rtsp-client.c:
9359 * gst/rtsp-server/rtsp-client.h:
9360 client: add method to filter managed sessions
9361 Add a method to filter the sessions managed by this client connection.
9362 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9364 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9366 * gst/rtsp-server/rtsp-client.c:
9367 * gst/rtsp-server/rtsp-client.h:
9368 client: remove _get_uri() method
9369 Remove the get_uri() method on the client. A client has no uri, the uri
9370 property is an internal property to manage the last cached media for
9373 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9375 * gst/rtsp-server/rtsp-media-factory.h:
9376 media-factory: fix typo
9378 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9380 * gst/rtsp-server/rtsp-media.c:
9381 rtsp-media: Do not leak the query in default_query_stop
9382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9384 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9386 * gst/rtsp-server/rtsp-media.c:
9387 media: don't unlock when conversion fails
9388 Don't unlock the state lock when conversion fails because it was not locked.
9390 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9392 * gst/rtsp-server/rtsp-media.c:
9393 * gst/rtsp-server/rtsp-media.h:
9394 Add query_position and query_stop vmethods to rtsp-media
9396 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9398 * gst/rtsp-server/rtsp-media.c:
9399 Fix typo in property install for rtsp-media's time-provider
9401 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9403 * gst/rtsp-server/rtsp-client.c:
9404 * gst/rtsp-server/rtsp-client.h:
9405 client: clean some variables
9406 Clean some variables and add some guards to _send_request()
9408 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9410 * gst/rtsp-server/rtsp-client.c:
9411 * gst/rtsp-server/rtsp-client.h:
9412 Add gst_rtsp_client_send_request API
9413 This makes it possible to send arbitrary messages to a client, such as
9414 SET_PARAMETER or GET_PARAMETER
9416 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9418 * gst/rtsp-server/rtsp-media.c:
9419 * gst/rtsp-server/rtsp-media.h:
9420 media: add _get_element() method
9421 Add method to get the element used when creating the media.
9422 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9424 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9426 * gst/rtsp-server/rtsp-media.c:
9429 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9431 * gst/rtsp-server/rtsp-stream.c:
9432 * gst/rtsp-server/rtsp-stream.h:
9433 stream: allow access to the rtp session
9434 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9436 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9438 * gst/rtsp-server/rtsp-stream.c:
9439 * gst/rtsp-server/rtsp-stream.h:
9440 dscp qos support in gst-rtsp-stream
9441 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9443 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9445 * tests/check/gst/rtspserver.c:
9447 Actually do what the comment says. Also keep the old code around, not sure what
9448 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9449 it currently doesn't.
9451 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9453 * gst/rtsp-server/rtsp-client.c:
9454 client: also watch newly created session
9455 When we newly created a session, start watching it immediately instead of
9456 on the next request.
9458 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9460 * tests/check/gst/client.c:
9461 tests: add unit test for new-session
9462 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9464 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9466 * gst/rtsp-server/rtsp-client.c:
9467 client: emit new-session when new session is created
9468 Only emit new-session when we created a new session for a client, not when a
9469 client picked up a previous session.
9470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9472 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9474 * gst/rtsp-server/rtsp-client.c:
9475 client: handle asterisk as path in requests
9476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9478 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9480 * gst/rtsp-server/rtsp-media.c:
9481 media: handle segment query format mismatch
9482 It's possible that the segment query returns with a different format than what
9483 we asked for, handle this case also.
9485 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9487 * gst/rtsp-server/rtsp-media.c:
9488 media: use segment stop in collect_media_stats
9489 Use segment stop instead of duration as range end point.
9490 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9492 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9494 * gst/rtsp-server/rtsp-media.c:
9495 * tests/check/gst/media.c:
9496 rtsp-media: Do not leak the element in take_pipeline
9497 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9499 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9501 * gst/rtsp-server/rtsp-client.c:
9502 * gst/rtsp-server/rtsp-client.h:
9503 rtsp-client: Make configure_client_transport virtual
9504 This patch makes configure_client_transport virtual. The functionality is
9505 needed to handle some weird clients sending multicast transport settings as url
9507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9509 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9511 * gst/rtsp-server/rtsp-client.c:
9512 * gst/rtsp-server/rtsp-client.h:
9513 rtsp-client: Make param_set and param_get virtual
9514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9516 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9518 * gst/rtsp-server/rtsp-client.c:
9519 * gst/rtsp-server/rtsp-media.c:
9520 * gst/rtsp-server/rtsp-media.h:
9521 media: convert_range replaces get_range_times
9522 get_range_times worked for handling UTC ranges for seeks, but we also
9523 need to convert back from NPT to the requested unit in
9524 get_range_string. convert_range is now used for both.
9525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9527 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9529 * gst/rtsp-server/rtsp-client.c:
9530 * gst/rtsp-server/rtsp-sdp.c:
9531 * gst/rtsp-server/rtsp-sdp.h:
9532 sdp: cleanup sdp info
9533 We don't need to pass the proto, we can more easily check a boolean.
9534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9536 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9538 * gst/rtsp-server/rtsp-sdp.c:
9539 use 0.0.0.0 or :: for c= line instead of server address
9541 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9543 * gst/rtsp-server/rtsp-client.c:
9544 use local address, not remote, in SDP
9545 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9547 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9550 Automatic update of common submodule
9551 From 098c0d7 to 01a7a46
9553 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9555 * gst/rtsp-server/rtsp-media.c:
9556 * gst/rtsp-server/rtsp-media.h:
9557 media: possibility to override range time conversion
9558 Make it possible to override the conversion from GstRTSPTimeRange to
9559 GstClockTimes, that is done before seeking on the media
9560 pipeline. Overriding can be useful for UTC ranges, where the default
9561 conversion gives nanoseconds since 1900.
9562 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9564 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9566 * gst/rtsp-server/rtsp-server.c:
9567 * gst/rtsp-server/rtsp-server.h:
9568 rtsp-server: Expose the use_client_settings API
9569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9571 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9573 * gst/rtsp-server/rtsp-client.c:
9574 * gst/rtsp-server/rtsp-stream.c:
9575 * gst/rtsp-server/rtsp-stream.h:
9576 rtspstream: handle both ipv4 and ipv6 clients
9577 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9579 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9581 * gst/rtsp-server/rtsp-sdp.c:
9582 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9583 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9584 We already have a way to place extra attributes in the SDP by using a string
9585 property with prefix x- or a- in the caps.
9587 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9589 * gst/rtsp-server/rtsp-sdp.c:
9590 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9591 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9592 We already have a way to place extra attributes in the SDP, just make a string
9593 property in the payloader with a- or x- prefix.
9595 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9597 * gst/rtsp-server/rtsp-sdp.c:
9598 rtsp: place a- and x- properties as attributes
9599 application/x-rtp has properties with a- and x- prefixes that should be
9600 placed as attributes in the SDP for the media instead of being added to the
9603 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9605 * examples/Makefile.am:
9606 * examples/test-video.c:
9607 example: add TLS example
9609 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9611 * gst/rtsp-server/rtsp-server.c:
9612 * gst/rtsp-server/rtsp-server.h:
9613 server: add support for TLS
9614 Add methods to set and get a TLS certificate.
9615 Add vmethod to configure a new connection. By default, configure the TLS
9616 certificate in a new connection if needed.
9618 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9620 * gst/rtsp-server/rtsp-server.c:
9621 * gst/rtsp-server/rtsp-server.h:
9622 server: remove accept_client vmethod
9623 This vmethod is not very useful so remove it.
9625 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9627 * gst/rtsp-server/rtsp-server.c:
9628 server: don't crash on NULL GError
9630 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
9632 * gst/rtsp-server/rtsp-session-pool.c:
9633 rtsp-session-pool: corrected session timeout detection
9634 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
9636 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9638 * gst/rtsp-server/rtsp-client.c:
9639 client: improve debug
9641 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 * gst/rtsp-server/rtsp-client.c:
9644 * gst/rtsp-server/rtsp-client.h:
9645 * gst/rtsp-server/rtsp-server.c:
9646 server: refactor connection setup
9647 Let the server accept the socket connection and construct a GstRTSPConnection
9648 from it. Remove the code from the client and let the client only deal with
9649 a fully configure GstRTSPConnection object.
9650 We will need this later when the server will configure the connection for
9653 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9655 * gst/rtsp-server/rtsp-stream.c:
9656 stream: keep the transport object alive
9657 Keep the transport object alive while we have it as qdata on the
9660 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
9662 * gst/rtsp-server/rtsp-client.c:
9663 * gst/rtsp-server/rtsp-server.c:
9664 rtsp-server: Do not crash on nmapping of server
9665 * generate error when gst_rtsp_connection_accept fails
9666 * do not stop accepting incoming connections because
9667 accepting a client fails
9668 https://bugzilla.gnome.org/show_bug.cgi?id=701072
9670 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
9672 * gst/rtsp-server/rtsp-client.c:
9673 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
9674 https://bugzilla.gnome.org/show_bug.cgi?id=700953
9676 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9678 * gst/rtsp-server/rtsp-sdp.c:
9679 rtsp-sdp: Parse framerate caps field and set SDP attribute
9680 The SDP attribute and its format is described in RFC4566.
9681 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9683 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
9685 * gst/rtsp-server/rtsp-sdp.c:
9686 rtsp-sdp: Parse width/height from caps and set SDP attribute
9687 The SDP attribute and its format is described in RFC6064.
9688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9690 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
9692 * gst/rtsp-server/rtsp-sdp.c:
9693 * tests/check/gst/client.c:
9694 rtsp-sdp: add bandwidth line
9695 https://bugzilla.gnome.org/show_bug.cgi?id=699220
9697 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9700 Automatic update of common submodule
9701 From 5edcd85 to 098c0d7
9703 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9705 * tests/check/gst/media.c:
9706 tests: add dynamic payloader prepare/unprepare check
9708 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9710 * gst/rtsp-server/rtsp-media.c:
9711 media: release lock when removing fakesink
9713 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9715 * gst/rtsp-server/rtsp-stream.c:
9716 stream: set elements to NULL before removing
9717 When removing a stream, set the elements to NULL first. This avoids
9718 element-is-not-in-NULL-state errors when we dispose the elements.
9720 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9723 Automatic update of common submodule
9724 From 3cb3d3c to 5edcd85
9726 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9728 * gst/rtsp-server/rtsp-media.c:
9729 * gst/rtsp-server/rtsp-media.h:
9730 media: listen to pad-removed signals
9731 Listen to the pad-removed signal and remove the stream associated with the
9733 Add signal to be notified of the removed pad.
9734 Remove the fakesink in unprepare()
9735 Fix signatures of the signal methods
9737 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9739 * examples/test-sdp.c:
9740 tests: add example of reusable pipelines
9742 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
9744 * gst/rtsp-server/rtsp-stream.c:
9745 * gst/rtsp-server/rtsp-stream.h:
9746 stream: add method to get the srcpad
9748 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9750 * tests/check/gst/media.c:
9751 check: add media prepare/unprepare test
9752 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9754 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
9756 * gst/rtsp-server/rtsp-media.c:
9757 media: disconnect from signal handlers in unprepare()
9758 We connected to the pad-added and no-more-pads signals in prepare() so
9759 we need to disconnect from them in unprepare().
9760 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9762 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
9764 * gst/rtsp-server/rtsp-media.c:
9765 media: don't free streams array
9766 Don't free the streams array in the unprepare() method, they were not
9768 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9770 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
9772 * gst/rtsp-server/rtsp-media.c:
9773 media: don't unref the pipeline in unprepare
9774 Unprepare() should undo what prepare() does. Because the pipeline is
9775 not created in prepare(), we should not unref it in unprepare()
9777 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
9779 * gst/rtsp-server/rtsp-stream.c:
9780 stream: clear session and caps for reuse
9781 Set the session and caps to NULL after unref otherwise we might unref
9783 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9785 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
9787 * gst/rtsp-server/rtsp-client.c:
9788 client: send out teardown signal before tearing down
9789 The advantage is that in the signal handler you get direct access to
9790 information about what streams are about to get torn down (in the
9791 GstRTSPClientState).
9792 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
9794 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
9796 * gst/rtsp-server/rtsp-client.c:
9797 * gst/rtsp-server/rtsp-client.h:
9798 client: expose connection
9799 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
9801 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
9804 Automatic update of common submodule
9805 From aed87ae to 3cb3d3c
9807 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9809 * gst/rtsp-server/rtsp-media.c:
9810 * gst/rtsp-server/rtsp-media.h:
9811 * gst/rtsp-server/rtsp-session-media.c:
9812 * gst/rtsp-server/rtsp-session-media.h:
9813 media: add method to get the base_time of the pipeline
9814 Together with a shared clock, this base-time could eventually be sent to
9815 the client so that it can reconstruct the exact running-time of the clock
9818 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9820 * gst/rtsp-server/Makefile.am:
9821 * gst/rtsp-server/rtsp-media.c:
9822 * gst/rtsp-server/rtsp-media.h:
9823 * gst/rtsp-server/rtsp-sdp.c:
9824 media: add GstNetTimeProvider support
9825 Add a property to let the media provide a GstNetTimeProvider for its clock.
9826 Make methods to get the clock and nettimeprovider
9827 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
9828 provider and also the current time of the clock. This should make it possible
9829 for (GStreamer) clients to slave their clock to the server clock.
9831 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
9834 Automatic update of common submodule
9835 From 04c7a1e to aed87ae
9837 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9839 * gst/rtsp-server/rtsp-media.c:
9840 media: wait for buffering to complete
9841 Wait for buffering to complete before changing the state to the target state.
9843 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9845 * gst/rtsp-server/rtsp-media.c:
9846 media: small cleanup
9848 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
9850 * tests/check/gst/rtspserver.c:
9851 tests: remove extra unref in test_setup_non_existing_stream
9852 The unref is not needed anymore, teardown runs without it.
9853 https://bugzilla.gnome.org/show_bug.cgi?id=696542
9855 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
9857 * tests/check/gst/rtspserver.c:
9858 tests: GSocketService cleanup in test_bind_already_in_use
9859 Use g_socket_service_stop so the rtspserver test stops listening for
9860 incoming connections in test_bind_already_in_use.
9861 https://bugzilla.gnome.org/show_bug.cgi?id=696541
9863 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
9865 * gst/rtsp-server/rtsp-media-factory.c:
9866 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
9867 Instead use a GWeakRef which is safe to use
9868 This is a known GLib bug, see:
9869 https://bugzilla.gnome.org/show_bug.cgi?id=667145
9871 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
9873 * gst/rtsp-server/rtsp-client.c:
9874 * gst/rtsp-server/rtsp-media.c:
9875 * gst/rtsp-server/rtsp-media.h:
9876 * gst/rtsp-server/rtsp-sdp.c:
9877 * tests/check/gst/media.c:
9878 * tests/check/gst/rtspserver.c:
9879 rtsp-media/client: Reply to PLAY request with same type of Range
9880 Remember the type of Range from the PLAY request and use the same type for
9883 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
9885 * gst/rtsp-server/rtsp-client.c:
9886 * gst/rtsp-server/rtsp-client.h:
9887 * tests/check/gst/client.c:
9888 rtsp-client: expose uri
9890 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
9892 * tests/check/gst/mediafactory.c:
9893 tests: Hold ref while creating second media
9894 To test if the media aren't shared, make sure we keep the first one while creating a second
9895 otherwise the same memory address may be reused.
9897 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
9900 configure: remove out-of-date comment
9902 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
9905 .gitignore: ignore more build files
9907 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
9909 * tests/check/Makefile.am:
9910 tests: use right _LIBS variable for gst-plugins-base libs
9912 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9914 * tests/check/Makefile.am:
9915 check: add librtp to libs
9917 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
9919 * tests/check/gst/rtspserver.c:
9920 tests: Add test to check selecting a port the server will send from
9922 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
9924 * tests/check/gst/rtspserver.c:
9925 tests: Make sure packets are actually received
9927 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9929 * gst/rtsp-server/rtsp-stream.c:
9930 stream: Select unicast address from pool if appropriate
9932 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
9934 * gst/rtsp-server/rtsp-stream.c:
9935 stream: Properties are always there in Gst 1.0
9937 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9939 * tests/check/gst/addresspool.c:
9940 tests: Add tests for unicast addresses in pool
9942 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
9944 * gst/rtsp-server/rtsp-address-pool.c:
9945 * tests/check/gst/addresspool.c:
9946 address-pool: Verify that multicast addresses are used for multicast and vice-versa
9948 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
9950 * docs/libs/gst-rtsp-server-sections.txt:
9951 * gst/rtsp-server/rtsp-address-pool.c:
9952 * gst/rtsp-server/rtsp-address-pool.h:
9953 * gst/rtsp-server/rtsp-stream.c:
9954 * tests/check/gst/addresspool.c:
9955 address-pool: Add unicast addresses
9957 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9960 * gst/rtsp-server/rtsp-server.c:
9961 * tests/check/gst/rtspserver.c:
9962 rtsp-server: Limit the number of threads per server instance
9963 If we exceed the maximum, just round robin the clients over the existing
9966 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
9968 * gst/rtsp-server/rtsp-server.c:
9969 rtsp-server: No need to store the GMainContext in the client context
9971 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
9973 * tests/check/gst/rtspserver.c:
9974 tests: Add test for client disconnection
9976 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9978 * tests/check/gst/rtspserver.c:
9979 tests: Test client and session timeouts with multiple threads
9981 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
9983 * gst/rtsp-server/rtsp-address-pool.c:
9984 * gst/rtsp-server/rtsp-auth.c:
9985 * gst/rtsp-server/rtsp-client.c:
9986 * gst/rtsp-server/rtsp-media-factory-uri.c:
9987 * gst/rtsp-server/rtsp-media-factory.c:
9988 * gst/rtsp-server/rtsp-media.c:
9989 * gst/rtsp-server/rtsp-mount-points.c:
9990 * gst/rtsp-server/rtsp-server.c:
9991 * gst/rtsp-server/rtsp-session-media.c:
9992 * gst/rtsp-server/rtsp-session-pool.c:
9993 * gst/rtsp-server/rtsp-session.c:
9994 Document locking and its order
9996 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
9998 * tests/check/gst/rtspserver.c:
9999 tests: Test that slow DESCRIBE don't block other clients
10001 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10003 * tests/check/gst/client.c:
10004 tests: Add tests for client-requested multicast address
10006 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10008 * docs/libs/gst-rtsp-server-sections.txt:
10009 docs: Put the various functions in the right sections
10011 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10013 * docs/libs/gst-rtsp-server-docs.sgml:
10014 * docs/libs/gst-rtsp-server-sections.txt:
10015 * gst/rtsp-server/rtsp-address-pool.c:
10016 * gst/rtsp-server/rtsp-address-pool.h:
10017 docs: Generate docs for GstRTSPAddressPool
10019 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10021 * gst/rtsp-server/rtsp-client.c:
10022 * gst/rtsp-server/rtsp-stream.c:
10023 * gst/rtsp-server/rtsp-stream.h:
10024 client: Check client provided addresses against the address pool
10026 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10028 * gst/rtsp-server/rtsp-address-pool.c:
10029 * gst/rtsp-server/rtsp-address-pool.h:
10030 * tests/check/gst/addresspool.c:
10031 address-pool: Add API to request a specific address from the pool
10032 Also add relevant unit tests.
10034 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10036 * tests/check/gst/mediafactory.c:
10037 tests: Check the passing around of a RTSPAddressPool
10038 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10039 way down to the stream.
10041 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10043 * tests/check/gst/addresspool.c:
10044 tests: Add more tests for the address pool
10046 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10048 * gst/rtsp-server/rtsp-address-pool.c:
10049 address-pool: Fix off by one error
10050 When splitting a port range, the port after a skip is not part of range.
10052 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10055 Automatic update of common submodule
10056 From 2de221c to 04c7a1e
10058 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10061 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10062 AM_CONFIG_HEADER was removed in automake 1.13
10063 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10065 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10068 Automatic update of common submodule
10069 From a942293 to 2de221c
10071 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10073 * gst/rtsp-server/rtsp-client.c:
10074 client: make sure the watch exists while sending data
10075 Protect the send_func with a lock. This allows us to wait for sending
10076 to complete before changing the send_func and user_data. We add an
10077 extra ref to the watch to make sure that it remains valid during
10079 When closing the connection, set the send_func to NULL
10080 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10082 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10084 * tests/check/Makefile.am:
10085 tests: use GST_*_1_0 environment variables everywhere
10086 The _1_0 suffixed environment variables override the
10087 non-suffixed ones, so if we're in an environment that
10088 sets the _1_0 suffixed ones, such as jhbuild, we need
10089 to set those to make sure ours actually always get
10092 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10095 Automatic update of common submodule
10096 From acb04d9 to a942293
10098 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10100 * gst/rtsp-server/rtsp-client.c:
10101 rtsp-client: set the client backlog
10102 Set the client backlog to a reasonable default
10104 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10106 * gst/rtsp-server/rtsp-media.c:
10107 rtsp-media: Make the element a constructor parameter
10108 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10110 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10112 * docs/libs/Makefile.am:
10113 docs: Link with gcov library when gcov is enabled
10114 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10116 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10118 * gst/rtsp-server/rtsp-media.c:
10119 media: match prepare with unprepare
10120 Really unprepare when there were an equal amount of prepare calls.
10122 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10124 * gst/rtsp-server/rtsp-media.c:
10125 media: media has to be unprepared in finalize
10126 Because unprepare takes away the last ref on the media.
10128 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10130 * gst/rtsp-server/rtsp-client.c:
10131 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10132 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10133 We can't use the refcount to trigger unprepare because it is the unprepare call
10134 that removes the last refcount after all messages are consumed. What we should
10135 probably do is make a prepared refcount and only unprepare when the refcount
10138 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10140 * gst/rtsp-server/rtsp-media.c:
10141 media: let the source unref the last media ref
10142 the last ref to the media is held by the source so we don't need to add more ref
10143 and unrefs, we simply destroy the media when the source is gone.
10145 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10147 * gst/rtsp-server/rtsp-media.c:
10148 media: improve debug
10150 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10152 * gst/rtsp-server/rtsp-media.c:
10154 Make sure we are in the right state when collecting the position and duration.
10155 Only make ourselves PREPARED when we were previously PREPARING.
10157 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10159 * gst/rtsp-server/rtsp-media.c:
10160 media: use g_object_ref/unref for GObjects
10162 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10164 * gst/rtsp-server/rtsp-client.c:
10165 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10166 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10167 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10168 isn't being used anymore.
10170 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10172 * gst/rtsp-server/rtsp-media.c:
10173 Fix compiler warning
10175 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10177 * gst/rtsp-server/rtsp-media-factory-uri.c:
10178 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10180 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10182 * gst/rtsp-server/rtsp-session-media.h:
10185 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10187 * gst/rtsp-server/rtsp-media.c:
10188 * tests/check/gst/media.c:
10189 media: avoid element leak
10191 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10193 * gst/rtsp-server/rtsp-media.c:
10194 media: require an element in media constructor
10196 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10198 * gst/rtsp-server/rtsp-client.c:
10199 Revert "client: TEARDOWN brings that state to Init again"
10200 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10201 The object is already disposed, there is no point in setting the state.
10203 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10205 * gst/rtsp-server/rtsp-client.c:
10206 client: TEARDOWN brings that state to Init again
10208 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10210 * docs/libs/gst-rtsp-server-sections.txt:
10211 * examples/test-auth.c:
10212 * gst/rtsp-server/rtsp-auth.c:
10213 * gst/rtsp-server/rtsp-auth.h:
10214 * gst/rtsp-server/rtsp-client.c:
10215 * gst/rtsp-server/rtsp-client.h:
10216 * gst/rtsp-server/rtsp-media-factory-uri.c:
10217 * gst/rtsp-server/rtsp-media-factory-uri.h:
10218 * gst/rtsp-server/rtsp-media-factory.c:
10219 * gst/rtsp-server/rtsp-media-factory.h:
10220 * gst/rtsp-server/rtsp-media.c:
10221 * gst/rtsp-server/rtsp-media.h:
10222 * gst/rtsp-server/rtsp-mount-points.c:
10223 * gst/rtsp-server/rtsp-mount-points.h:
10224 * gst/rtsp-server/rtsp-sdp.c:
10225 * gst/rtsp-server/rtsp-server.c:
10226 * gst/rtsp-server/rtsp-server.h:
10227 * gst/rtsp-server/rtsp-session-media.c:
10228 * gst/rtsp-server/rtsp-session-media.h:
10229 * gst/rtsp-server/rtsp-session-pool.c:
10230 * gst/rtsp-server/rtsp-session-pool.h:
10231 * gst/rtsp-server/rtsp-session.c:
10232 * gst/rtsp-server/rtsp-session.h:
10233 * gst/rtsp-server/rtsp-stream-transport.c:
10234 * gst/rtsp-server/rtsp-stream-transport.h:
10235 * gst/rtsp-server/rtsp-stream.c:
10236 * gst/rtsp-server/rtsp-stream.h:
10237 * tests/check/gst/media.c:
10238 rtsp: make object details private
10239 Make all object details private
10240 Add methods to access private bits
10242 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10244 * tests/check/Makefile.am:
10245 * tests/check/gst/media.c:
10246 tests: add media tests
10248 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10250 * gst/rtsp-server/rtsp-media.c:
10251 media: check if prepared for some methods
10252 Check that the media object is prepared before doing seek and getting the
10253 current position etc.
10254 Add some g_return checks.
10256 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10258 * tests/check/Makefile.am:
10259 * tests/check/gst/mediafactory.c:
10260 tests: add mediafactory test
10262 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10264 * gst/rtsp-server/rtsp-stream.c:
10265 stream: improve debug
10267 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10269 * gst/rtsp-server/rtsp-media.c:
10270 * gst/rtsp-server/rtsp-media.h:
10271 media: unref pipeline in finalize to avoid leaking it
10273 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10275 * gst/rtsp-server/rtsp-media-factory-uri.c:
10276 * gst/rtsp-server/rtsp-media.c:
10277 rtsp: use gst_object_unref on GstObjects
10279 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10281 * gst/rtsp-server/rtsp-media-factory.c:
10282 media-factory: require an url
10284 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10286 * examples/test-uri.c:
10287 examples: fix include
10289 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10291 * gst/rtsp-server/rtsp-server.h:
10292 server: remove unused include
10294 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10296 * tests/check/Makefile.am:
10297 * tests/check/gst/mountpoints.c:
10298 tests: add test for mountpoints
10300 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10302 * gst/rtsp-server/rtsp-client.c:
10303 client: fix factory leak
10304 Keep the factory in the state object only for authorization checks and make
10305 sure we unref it on failure. Also don't keep invalid objects in the state
10308 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10310 * gst/rtsp-server/rtsp-mount-points.c:
10311 mounts: add g_return_if guards
10313 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10315 * tests/check/gst/client.c:
10316 tests: add more tests
10318 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10320 * gst/rtsp-server/rtsp-client.c:
10321 client: improve debug
10323 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10325 * gst/rtsp-server/rtsp-client.c:
10326 client: improve debug and fix leaks
10327 Cleanup the uri and session when there is a bad request.
10329 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10334 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10336 * tests/check/gst/client.c:
10337 test: add test for session in options request
10339 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10341 * gst/rtsp-server/rtsp-client.c:
10342 client: use 454 when session can't be found
10343 We should use 454 when a session can't be found because there was no session
10344 pool configured in the server. This is not a server configuration problem
10345 because the server on which the request is done might not be the same one that
10346 will keep the sessions for us and so it does not need to support sessions.
10348 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10350 * gst/rtsp-server/rtsp-client.c:
10351 client: only free connection when there is one
10352 It's possible that the client doesn't have a connection when we try to free it.
10354 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10356 * tests/check/Makefile.am:
10357 * tests/check/gst/client.c:
10358 tests: add unit test for the client object
10360 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10362 * gst/rtsp-server/rtsp-client.c:
10363 client: small cleanup
10365 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10367 * gst/rtsp-server/rtsp-client.h:
10368 client: remove unused include
10370 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10372 * gst/rtsp-server/rtsp-client.c:
10373 client: fix compilation
10375 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10377 * gst/rtsp-server/rtsp-client.c:
10378 client: call destroy without the lock
10380 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10382 * gst/rtsp-server/rtsp-client.c:
10383 * gst/rtsp-server/rtsp-client.h:
10384 client: make the client usable without a socket
10385 Make a method to let the client handle a message and a callback when the client
10386 wants us to send a response message back. This makes it possible to also use the
10387 client object without the sockets, which should make it easier to test.
10389 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10391 * gst/rtsp-server/rtsp-client.c:
10392 * gst/rtsp-server/rtsp-client.h:
10393 client: small cleanup
10395 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10397 * docs/libs/gst-rtsp-server-sections.txt:
10398 * gst/rtsp-server/rtsp-client.c:
10399 * gst/rtsp-server/rtsp-client.h:
10400 * gst/rtsp-server/rtsp-server.c:
10401 client: remove reference to server
10402 We don't need to keep a ref to the server
10404 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10406 * gst/rtsp-server/rtsp-client.c:
10407 * gst/rtsp-server/rtsp-client.h:
10408 client: add locking
10409 Also add some g_return_if()
10411 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10413 * gst/rtsp-server/rtsp-client.c:
10414 client: log more errors
10416 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10418 * gst/rtsp-server/rtsp-client.c:
10419 client: fix compilation
10421 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10423 * gst/rtsp-server/rtsp-client.c:
10424 * gst/rtsp-server/rtsp-client.h:
10425 client: add generic close-after-send support
10426 Add a property to send_response() to close the connection after the response has
10427 been sent to the client.
10429 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10432 * docs/libs/gst-rtsp-server-docs.sgml:
10433 * docs/libs/gst-rtsp-server-sections.txt:
10434 * docs/libs/gst-rtsp-server.types:
10435 * examples/test-auth.c:
10436 * examples/test-launch.c:
10437 * examples/test-mp4.c:
10438 * examples/test-multicast.c:
10439 * examples/test-multicast2.c:
10440 * examples/test-ogg.c:
10441 * examples/test-readme.c:
10442 * examples/test-sdp.c:
10443 * examples/test-uri.c:
10444 * examples/test-video.c:
10445 * gst/rtsp-server/Makefile.am:
10446 * gst/rtsp-server/rtsp-auth.h:
10447 * gst/rtsp-server/rtsp-client.c:
10448 * gst/rtsp-server/rtsp-client.h:
10449 * gst/rtsp-server/rtsp-media-mapping.c:
10450 * gst/rtsp-server/rtsp-media-mapping.h:
10451 * gst/rtsp-server/rtsp-mount-points.c:
10452 * gst/rtsp-server/rtsp-mount-points.h:
10453 * gst/rtsp-server/rtsp-server.c:
10454 * gst/rtsp-server/rtsp-server.h:
10455 * gst/rtsp-server/rtsp-session-media.c:
10456 * gst/rtsp-server/rtsp-session-pool.c:
10457 * gst/rtsp-server/rtsp-session-pool.h:
10458 * tests/check/gst/rtspserver.c:
10459 MediaMapping -> MountPoints
10460 Describes better what the object manages.
10462 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10465 configure: bump required version of -base
10467 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10469 * gst/rtsp-server/rtsp-media.c:
10472 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10474 * gst/rtsp-server/rtsp-media.c:
10475 * gst/rtsp-server/rtsp-media.h:
10476 media: support more Range formats
10477 Use the new -base methods to convert the Range string into a seek start and stop
10480 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10482 * examples/test-launch.c:
10483 examples: fix whitespace
10485 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10487 * examples/test-auth.c:
10488 test-auth: add example of how to remove sessions
10489 Add an example of the session filter api.
10491 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10493 * examples/test-uri.c:
10494 test-uri: remove mapping example
10496 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10498 * examples/test-uri.c:
10499 test-uri: fix callback signature
10501 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10503 * gst/rtsp-server/rtsp-media-factory.c:
10504 factory: keep ref to factory while media active
10505 While the media from a factory is alive, keep a ref to the factory.
10506 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10508 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10510 * gst/rtsp-server/rtsp-media-factory-uri.c:
10511 factory-uri: add some debug
10513 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10515 * gst/rtsp-server/rtsp-stream.c:
10516 stream: set udp sources to PLAYING
10517 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10518 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10520 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10522 * gst/rtsp-server/rtsp-media-factory-uri.c:
10523 factory-uri: take ref to factory
10524 Take a ref to the factory that we place in our list.
10526 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10528 * tests/Makefile.am:
10529 * tests/test-reuse.c:
10530 test: add test for server reuse
10531 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10533 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10535 * gst/rtsp-server/rtsp-server.c:
10536 server: start and stop multiple times
10537 Stop listening on the RTSP port when the GSource is removed, so clients
10538 can't connect and the server can be started again.
10539 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10541 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10543 * gst/rtsp-server/rtsp-server.c:
10544 server: fix small leak
10546 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10548 * gst/rtsp-server/rtsp-media.c:
10549 media: unref source in finish_unprepare
10550 The source is created in prepare, unref it in finish_unprepare.
10551 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10553 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10555 * gst/rtsp-server/rtsp-client.c:
10556 * gst/rtsp-server/rtsp-media.c:
10557 rtsp-media: remove bus watch before finalizing
10558 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10559 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10560 the GDestroyNotify function.
10561 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10562 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10563 gst_rtsp_media_unprepare before unreffing the media.
10564 This way, the bus watch will be removed before the media is finalized.
10565 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10567 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10569 * gst/rtsp-server/rtsp-client.c:
10570 * gst/rtsp-server/rtsp-client.h:
10571 client: wait until the TEARDOWN response is sent to close the connection
10572 Responses can be sent async so we need to wait until the TEARDOWN response has
10573 been written before we close the connection to the client. This avoids the risk
10574 of writing/polling closed sockets.
10575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10577 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10579 * gst/rtsp-server/rtsp-stream.c:
10580 rtsp-stream: plug socket leak
10581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10583 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10586 Automatic update of common submodule
10587 From 6bb6951 to a72faea
10589 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10591 * gst/rtsp-server/rtsp-media-factory-uri.c:
10592 rtsp-server: don't use deprecated API
10594 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10596 * gst/rtsp-server/rtsp-client.c:
10597 rtsp-client: fix unused-but-set-variable compiler warning
10598 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10600 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10603 * docs/libs/gst-rtsp-server-sections.txt:
10604 * gst/rtsp-server/rtsp-client.c:
10607 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10609 * examples/Makefile.am:
10610 * examples/test-multicast2.c:
10611 examples: add another multicast example
10612 Add an example for how to configure separate multicast ranges for each media
10615 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10617 * examples/test-multicast.c:
10620 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10622 * gst/rtsp-server/rtsp-client.c:
10623 * gst/rtsp-server/rtsp-media.c:
10624 * gst/rtsp-server/rtsp-session-media.c:
10625 * gst/rtsp-server/rtsp-session-media.h:
10626 * gst/rtsp-server/rtsp-stream-transport.c:
10627 * gst/rtsp-server/rtsp-stream-transport.h:
10628 stream: use the address managed by the stream
10629 Use the address managed by the stream for multicast. This allows us to have 1
10630 multicast address for each stream.
10631 Because the address is now managed by the stream we don't have to pass it around
10633 Set the address pool on the streams.
10635 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10637 * gst/rtsp-server/rtsp-client.c:
10638 * gst/rtsp-server/rtsp-media.c:
10639 * gst/rtsp-server/rtsp-stream.c:
10640 rtsp: improve debug
10642 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10644 * gst/rtsp-server/rtsp-media.c:
10645 * gst/rtsp-server/rtsp-media.h:
10646 media: add signal for new streams
10647 This allows applications to listen for new streams and configure properties on
10648 them, like the address pool.
10650 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10652 * gst/rtsp-server/rtsp-media.c:
10653 media: configure address pool in new streams
10655 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10657 * gst/rtsp-server/rtsp-stream.c:
10658 * gst/rtsp-server/rtsp-stream.h:
10659 stream: add methods to deal with address pool
10660 Add methods to get and set the address pool for the stream
10661 Add method to allocate and get the multicast addresses for this stream.
10663 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10665 * docs/libs/gst-rtsp-server-sections.txt:
10666 * gst/rtsp-server/rtsp-media.c:
10667 * gst/rtsp-server/rtsp-media.h:
10668 media: remove MTU property
10669 It is a stream property
10671 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10673 * gst/rtsp-server/rtsp-client.c:
10674 client: set blocksize only on stream
10675 Set the blocksize only on the current stream.
10677 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10679 * gst/rtsp-server/rtsp-stream.c:
10680 stream: share src and sink sockets
10681 the allocated socket is in the used-socket property, not socket.
10683 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10685 * gst/rtsp-server/rtsp-address-pool.c:
10686 * gst/rtsp-server/rtsp-address-pool.h:
10687 * gst/rtsp-server/rtsp-client.c:
10688 * gst/rtsp-server/rtsp-session-media.c:
10689 * gst/rtsp-server/rtsp-session-media.h:
10690 * gst/rtsp-server/rtsp-stream-transport.c:
10691 * gst/rtsp-server/rtsp-stream-transport.h:
10692 * tests/check/gst/addresspool.c:
10693 rtsp: make address-pool return an address object
10694 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
10695 store more info in the structure and allows us to more easily return the address
10696 to the right pool when no longer needed.
10697 Pass the address to the StreamTransport so that we can return it to the pool
10698 when the stream transport is freed or changed.
10700 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10702 * examples/Makefile.am:
10703 * examples/test-multicast.c:
10704 examples: add multicast example
10705 Show how to set up the multicast address pool so that media can be
10706 server with multicast.
10708 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10710 * gst/rtsp-server/rtsp-client.c:
10711 * gst/rtsp-server/rtsp-media-factory.c:
10712 * gst/rtsp-server/rtsp-media-factory.h:
10713 * gst/rtsp-server/rtsp-media.c:
10714 * gst/rtsp-server/rtsp-media.h:
10715 rtsp: use AddressPool
10716 Remove the multicast_group property.
10717 Use the configured addresspool to allocate multicast addresses.
10719 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10721 * gst/rtsp-server/rtsp-address-pool.c:
10722 * gst/rtsp-server/rtsp-address-pool.h:
10723 address-pool: add clear method
10725 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10727 * gst/rtsp-server/rtsp-address-pool.c:
10728 address-pool: small cleanups
10730 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10732 * tests/check/Makefile.am:
10733 * tests/check/gst/addresspool.c:
10734 tests: add addresspool unit test
10736 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10738 * gst/rtsp-server/Makefile.am:
10739 * gst/rtsp-server/rtsp-address-pool.c:
10740 * gst/rtsp-server/rtsp-address-pool.h:
10741 address-pool: add object to manage multicast addresses
10742 Make an object that can manage a rage of multicast addresses and ports.
10744 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10746 * gst/rtsp-server/rtsp-server.c:
10747 server: set default max-threads property
10749 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10751 * gst/rtsp-server/rtsp-media.c:
10752 media: wait for concurrent _prepare
10753 If a prepare is busy, wait for the result.
10755 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10757 * gst/rtsp-server/rtsp-media.c:
10758 media: add lock around message handler
10759 We don't want to dispatch messages while we are still processing the result of
10762 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10764 * gst/rtsp-server/rtsp-media.c:
10765 * gst/rtsp-server/rtsp-media.h:
10766 media: add lock to protect state changes
10768 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10770 * gst/rtsp-server/rtsp-stream.c:
10771 * gst/rtsp-server/rtsp-stream.h:
10772 stream: add locking
10774 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10776 * gst/rtsp-server/rtsp-stream-transport.c:
10777 * gst/rtsp-server/rtsp-stream-transport.h:
10778 * gst/rtsp-server/rtsp-stream.c:
10779 stream-transport: add keep-alive method
10781 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10783 * gst/rtsp-server/rtsp-stream-transport.c:
10784 * gst/rtsp-server/rtsp-stream-transport.h:
10785 * gst/rtsp-server/rtsp-stream.c:
10786 stream-transport: add method to handle RTP/RTCP
10787 Call new methods instead of poking into the structures directly.
10789 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10791 * gst/rtsp-server/rtsp-session-media.c:
10792 * gst/rtsp-server/rtsp-session-media.h:
10793 session-media: add locking
10795 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10797 * gst/rtsp-server/rtsp-session.c:
10798 * gst/rtsp-server/rtsp-session.h:
10799 session: add locking
10801 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10803 * gst/rtsp-server/rtsp-server.c:
10804 server: free old socket
10806 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10808 * gst/rtsp-server/rtsp-media-mapping.c:
10809 * gst/rtsp-server/rtsp-media-mapping.h:
10810 mapping: add locking
10812 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10814 * gst/rtsp-server/rtsp-media-factory.c:
10815 media-factory: add locking
10817 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10819 * gst/rtsp-server/rtsp-auth.c:
10820 * gst/rtsp-server/rtsp-auth.h:
10823 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10825 * gst/rtsp-server/rtsp-server.c:
10826 * gst/rtsp-server/rtsp-server.h:
10827 server: add max-thread property
10829 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10831 * gst/rtsp-server/rtsp-server.c:
10832 * gst/rtsp-server/rtsp-server.h:
10833 server: use a threadpool for the mainloops
10835 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10837 * gst/rtsp-server/rtsp-client.c:
10838 * gst/rtsp-server/rtsp-client.h:
10839 client: rename method
10840 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
10841 don't really create the client from the socket, we use the socket for the
10844 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10846 * gst/rtsp-server/rtsp-client.c:
10847 * gst/rtsp-server/rtsp-client.h:
10848 * gst/rtsp-server/rtsp-server.c:
10849 server: rework maincontext handling in clients
10850 Make a separate method to attach a client to a MainContext.
10851 Let the server decide in what GMainContext the client will operate and give this
10852 context to the client in attach. Then the server can later decide to use a
10853 separate thread for each client or just use the mainthread.
10855 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10857 * gst/rtsp-server/rtsp-client.c:
10858 * gst/rtsp-server/rtsp-session.c:
10859 * gst/rtsp-server/rtsp-session.h:
10860 session: move session header code in session object
10862 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10866 * examples/test-auth.c:
10867 * examples/test-launch.c:
10868 * examples/test-mp4.c:
10869 * examples/test-ogg.c:
10870 * examples/test-readme.c:
10871 * examples/test-sdp.c:
10872 * examples/test-uri.c:
10873 * examples/test-video.c:
10874 * gst/rtsp-server/rtsp-auth.c:
10875 * gst/rtsp-server/rtsp-auth.h:
10876 * gst/rtsp-server/rtsp-client.c:
10877 * gst/rtsp-server/rtsp-client.h:
10878 * gst/rtsp-server/rtsp-media-factory-uri.c:
10879 * gst/rtsp-server/rtsp-media-factory-uri.h:
10880 * gst/rtsp-server/rtsp-media-factory.c:
10881 * gst/rtsp-server/rtsp-media-factory.h:
10882 * gst/rtsp-server/rtsp-media-mapping.c:
10883 * gst/rtsp-server/rtsp-media-mapping.h:
10884 * gst/rtsp-server/rtsp-media.c:
10885 * gst/rtsp-server/rtsp-media.h:
10886 * gst/rtsp-server/rtsp-params.c:
10887 * gst/rtsp-server/rtsp-params.h:
10888 * gst/rtsp-server/rtsp-sdp.c:
10889 * gst/rtsp-server/rtsp-sdp.h:
10890 * gst/rtsp-server/rtsp-server.c:
10891 * gst/rtsp-server/rtsp-server.h:
10892 * gst/rtsp-server/rtsp-session-media.c:
10893 * gst/rtsp-server/rtsp-session-media.h:
10894 * gst/rtsp-server/rtsp-session-pool.c:
10895 * gst/rtsp-server/rtsp-session-pool.h:
10896 * gst/rtsp-server/rtsp-session.c:
10897 * gst/rtsp-server/rtsp-session.h:
10898 * gst/rtsp-server/rtsp-stream-transport.c:
10899 * gst/rtsp-server/rtsp-stream-transport.h:
10900 * gst/rtsp-server/rtsp-stream.c:
10901 * gst/rtsp-server/rtsp-stream.h:
10902 * tests/check/gst/rtspserver.c:
10903 * tests/test-cleanup.c:
10906 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10908 * gst/rtsp-server/rtsp-media.c:
10909 * gst/rtsp-server/rtsp-session-media.c:
10910 * gst/rtsp-server/rtsp-session.c:
10911 rtsp-server: added annotations to indicate type of ownership transfer of return values
10912 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10914 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
10917 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
10919 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10922 * bindings/Makefile.am:
10923 * bindings/vala/Makefile.am:
10924 * bindings/vala/gst-rtsp-server-0.10.deps:
10925 * bindings/vala/gst-rtsp-server-0.10.vapi:
10926 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10927 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10928 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10929 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10930 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10932 bindings: remove vala bindings
10933 They'll be reunited with the other GStreamer bindings
10934 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10936 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10938 * gst/rtsp-server/rtsp-client.c:
10939 * gst/rtsp-server/rtsp-session-media.c:
10940 * gst/rtsp-server/rtsp-session-media.h:
10941 * gst/rtsp-server/rtsp-stream-transport.c:
10942 * gst/rtsp-server/rtsp-stream-transport.h:
10943 rtsp: only create transport when needed
10944 Only create the StreamTransport when configured.
10946 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10948 * gst/rtsp-server/rtsp-client.c:
10949 client: small cleanup
10951 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10953 * gst/rtsp-server/rtsp-client.c:
10954 * gst/rtsp-server/rtsp-client.h:
10955 * gst/rtsp-server/rtsp-stream-transport.c:
10956 * gst/rtsp-server/rtsp-stream-transport.h:
10957 rtsp: refactor configuration of transport
10958 Move the configuration of the transport to a place where it makes
10961 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10963 * gst/rtsp-server/rtsp-client.c:
10964 client: refactor transport parsing
10966 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10968 * gst/rtsp-server/rtsp-client.c:
10969 client: refuse to change the MTU on shared media
10970 If we change the MTU of chared media, it changes for all clients.
10971 We don't want to set the MTU to something large for clients that
10974 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10976 * examples/test-mp4.c:
10977 * gst/rtsp-server/rtsp-media.c:
10978 small fixes to docs and debug
10980 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10982 * gst/rtsp-server/rtsp-stream.c:
10983 stream: transports must already have been removed
10985 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10987 * gst/rtsp-server/rtsp-media.c:
10988 * gst/rtsp-server/rtsp-stream.c:
10989 * gst/rtsp-server/rtsp-stream.h:
10990 stream: improve join and leave of the pipeline
10992 Do the cleanup properly
10995 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10997 * gst/rtsp-server/rtsp-media.c:
10998 media: move unprepare below default implementation
10999 Makes it easier to find the default implementation
11001 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11003 * gst/rtsp-server/rtsp-media.c:
11004 media: signal unprepared when we actually finish
11006 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11008 * gst/rtsp-server/rtsp-media.c:
11009 media: no need to unlock, unprepare does that when needed
11011 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11013 * docs/libs/gst-rtsp-server-sections.txt:
11014 * gst/rtsp-server/rtsp-media-factory.h:
11015 * gst/rtsp-server/rtsp-media-mapping.c:
11016 * gst/rtsp-server/rtsp-media.h:
11017 * gst/rtsp-server/rtsp-params.c:
11018 * gst/rtsp-server/rtsp-server.c:
11019 * gst/rtsp-server/rtsp-session-pool.h:
11020 * gst/rtsp-server/rtsp-session.c:
11021 * gst/rtsp-server/rtsp-session.h:
11022 * gst/rtsp-server/rtsp-stream-transport.h:
11023 * gst/rtsp-server/rtsp-stream.h:
11026 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11028 * gst/rtsp-server/rtsp-client.c:
11029 * gst/rtsp-server/rtsp-media-mapping.h:
11030 * gst/rtsp-server/rtsp-media.c:
11031 * gst/rtsp-server/rtsp-media.h:
11032 * gst/rtsp-server/rtsp-server.h:
11033 * gst/rtsp-server/rtsp-stream.c:
11034 * gst/rtsp-server/rtsp-stream.h:
11035 rtsp: fix MTU setting
11036 Fix setting of the MTU. There is no need for a vmethod.
11038 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11043 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11046 configure: bump version number after refactoring
11048 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11050 * gst/rtsp-server/Makefile.am:
11051 * gst/rtsp-server/rtsp-client.c:
11052 * gst/rtsp-server/rtsp-client.h:
11053 * gst/rtsp-server/rtsp-media-factory-uri.c:
11054 * gst/rtsp-server/rtsp-media-factory.c:
11055 * gst/rtsp-server/rtsp-media-factory.h:
11056 * gst/rtsp-server/rtsp-media.c:
11057 * gst/rtsp-server/rtsp-media.h:
11058 * gst/rtsp-server/rtsp-sdp.c:
11059 * gst/rtsp-server/rtsp-session-media.c:
11060 * gst/rtsp-server/rtsp-session-media.h:
11061 * gst/rtsp-server/rtsp-session.c:
11062 * gst/rtsp-server/rtsp-session.h:
11063 * gst/rtsp-server/rtsp-stream-transport.c:
11064 * gst/rtsp-server/rtsp-stream-transport.h:
11065 * gst/rtsp-server/rtsp-stream.c:
11066 * gst/rtsp-server/rtsp-stream.h:
11067 rtsp: massive refactoring
11068 Make GObjects from the remaining simple structures.
11069 Remove GstRTSPSessionStream, it's not needed.
11070 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11071 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11072 a GstRTSPStream should be transported to a client.
11073 Rename GstRTSPMediaFactory::get_element -> create_element because that
11074 more accurately describes what it does.
11075 Make nice methods instead of poking in the structures.
11076 Move some methods inside the relevant object source code.
11077 Use GPtrArray to store objects instead of plain arrays, it is more
11078 natural and allows us to more easily clean up.
11079 Move the allocation of udp ports to the Stream object. The Stream object
11080 contains the elements needed to stream the media to a client.
11081 Improve the prepare and unprepare methods. Unprepare should now undo
11082 everything prepare did. Improve also async unprepare when doing EOS on
11083 shutdown. Make sure we always unprepare correctly.
11085 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11087 * gst/rtsp-server/rtsp-client.c:
11088 rtsp-client: Unref server address clients connected to
11089 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11091 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11093 * gst/rtsp-server/rtsp-server.c:
11094 rtsp-server: don't ref server socket if it is NULL
11095 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11096 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11098 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11100 * tests/check/Makefile.am:
11101 tests: Add libgio link dependency
11102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11104 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11106 * gst/rtsp-server/rtsp-media-mapping.c:
11107 * gst/rtsp-server/rtsp-media-mapping.h:
11108 rtsp-media-mapping: rename find_media vfunc to find_factory
11109 The virtual method and class method should have the same name
11110 so it is correctly represented in GIR file
11111 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11113 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11115 * gst/rtsp-server/rtsp-auth.c:
11116 * gst/rtsp-server/rtsp-client.c:
11117 * gst/rtsp-server/rtsp-media-factory-uri.c:
11118 * gst/rtsp-server/rtsp-media-factory.c:
11119 * gst/rtsp-server/rtsp-media-mapping.c:
11120 * gst/rtsp-server/rtsp-media.c:
11121 * gst/rtsp-server/rtsp-server.c:
11122 * gst/rtsp-server/rtsp-session-pool.c:
11123 * gst/rtsp-server/rtsp-session.c:
11124 rtsp-server: fixed comments and GIR annotations
11125 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11127 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11129 * gst/rtsp-server/rtsp-media-mapping.c:
11130 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11132 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11134 * gst/rtsp-server/rtsp-server.c:
11135 rtsp-server: allow binding on port 0 (binds on a random port)
11137 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11139 * gst/rtsp-server/rtsp-server.c:
11140 * gst/rtsp-server/rtsp-server.h:
11141 rtsp-server: add bound-port property
11142 bound-port can be used to retrieve the port number when the server is bound on
11143 port 0, which binds on a random port.
11145 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11147 * gst/rtsp-server/rtsp-media-factory.c:
11148 * gst/rtsp-server/rtsp-media-factory.h:
11149 rtsp-media-factory: make ::get_element overridable by GI bindings
11150 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11151 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11152 as the invoker for ::get_element(), making it overridable by GI generated
11155 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11157 * gst/rtsp-server/rtsp-media-factory-uri.c:
11158 rtsp-media-factory-uri: don't autoplug parsers in a loop
11159 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11162 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11164 * gst/rtsp-server/Makefile.am:
11165 Explicitly link against gio. Fix link error on mac.
11167 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11169 * gst/rtsp-server/rtsp-session.c:
11170 session: add ttl to the transport header in SETUP
11171 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11173 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11175 * gst/rtsp-server/rtsp-client.c:
11176 * gst/rtsp-server/rtsp-client.h:
11177 * gst/rtsp-server/rtsp-media.c:
11178 client: Use client transport settings for multicast if allowed.
11179 This patch makes it possible for the client to send transport settings for
11180 multicast (destination && ttl). Client settings must be explicitly allowed or
11181 the server will use its own settings.
11182 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11184 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11187 Automatic update of common submodule
11188 From 6c0b52c to 6bb6951
11190 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11192 * gst/rtsp-server/rtsp-client.c:
11193 rtsp-client: do not destroy the rtsp watch
11194 Don't destroy the client watch while dispatching. The rtsp watch is
11195 automatically destroyed after the rtsp watch function closed() has
11197 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11199 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11202 Automatic update of common submodule
11203 From 4f962f7 to 6c0b52c
11205 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11207 * gst/rtsp-server/rtsp-media.c:
11208 media: fix check for seekability
11210 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11212 * gst/rtsp-server/rtsp-client.c:
11213 client: use more GIO
11214 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11216 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11218 * gst/rtsp-server/rtsp-server.c:
11219 server: remove obsolete includes
11221 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11223 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11224 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11225 be available in "on_new_ssrc". The transports are added in
11226 gst_rtsp_media_set_state when going to PLAYING state. However,
11227 "on_new_ssrc" might be called before this happens.
11228 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11230 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11232 * gst/rtsp-server/rtsp-client.c:
11233 * gst/rtsp-server/rtsp-client.h:
11234 rtsp-client: add signals for rtsp requests (fixes #683287)
11236 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11238 * gst/rtsp-server/rtsp-client.c:
11239 * gst/rtsp-server/rtsp-client.h:
11240 add new-session signal to rtsp-client (fixes #683058)
11242 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11245 Automatic update of common submodule
11246 From 668acee to 4f962f7
11248 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11250 * gst/rtsp-server/rtsp-server.c:
11251 * tests/check/gst/rtspserver.c:
11252 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11253 Do not assume that *error is set in g_socket_address_enumerator_next.
11254 Added test_bind_already_in_use unit-test.
11255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11257 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11260 Automatic update of common submodule
11261 From 94ccf4c to 668acee
11263 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11265 * gst/rtsp-server/rtsp-client.c:
11266 * gst/rtsp-server/rtsp-client.h:
11267 rtsp-client: make create_sdp virtual method
11268 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11270 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11273 Automatic update of common submodule
11274 From 98e386f to 94ccf4c
11276 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11278 * gst/rtsp-server/rtsp-client.c:
11281 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11283 * gst/rtsp-server/rtsp-client.c:
11284 * gst/rtsp-server/rtsp-client.h:
11285 * gst/rtsp-server/rtsp-server.c:
11286 * gst/rtsp-server/rtsp-server.h:
11287 rtsp-server: use an existing socket to establish HTTP tunnel
11288 Make it possible to transfer a socket from an HTTP server to be used as
11289 an RTSP over HTTP tunnel.
11291 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11293 * gst/rtsp-server/rtsp-client.c:
11294 * gst/rtsp-server/rtsp-media.c:
11295 * gst/rtsp-server/rtsp-media.h:
11296 rtsp: Handle the blocksize parameter
11297 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11299 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11301 * tests/check/Makefile.am:
11302 * tests/check/gst/rtspserver.c:
11303 Have unit test get header from source dir, not installed dir
11304 This makes compilation of unit tests work in a build directory other
11305 than the source directory.
11306 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11308 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11310 * gst/rtsp-server/rtsp-media.c:
11311 rtsp-media: update for gst_element_make_from_uri() changes
11313 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11316 * tests/Makefile.am:
11317 * tests/check/Makefile.am:
11318 * tests/check/gst/rtspserver.c:
11319 rtsp: add unit test
11320 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11322 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11324 * gst/rtsp-server/rtsp-media.c:
11325 rtsp-media: don't collect media stats when going to NULL
11326 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11328 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11330 * gst/rtsp-server/rtsp-client.c:
11331 client: don't leak transports
11333 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11335 * gst/rtsp-server/rtsp-client.c:
11336 rtsp-client: free transport on no_stream in SETUP handler
11338 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11340 * gst/rtsp-server/rtsp-client.c:
11341 rtsp-client: changed session media iteration
11342 In client_unlink_session: now don't iterate in session->medias
11343 list where items are removed by gst_rtsp_session_release_media.
11344 Instead, repeatedly remove the first item.
11346 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11348 * gst/rtsp-server/rtsp-client.c:
11349 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11350 GstRTSPSessionMedia is not a GObject type. When the
11351 GstRTSPSession is freed, it will free the media.
11353 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11355 * gst/rtsp-server/rtsp-media-factory.c:
11356 factory: plug pad leak in collect_streams
11357 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11358 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11359 will take one reference, and the other reference will otherwise
11360 give a memory leak.
11362 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11365 configure: suppress some warnings when debug is disabled
11366 Warnings about unused variables should be suppressed if core has the
11367 debug system disabled.
11368 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11370 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11372 * docs/libs/Makefile.am:
11373 docs: fix build in uninstalled setup
11374 Include gst-plugins-base libs properly.
11376 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11378 * docs/libs/gst-rtsp-server.types:
11379 docs: include headers defining rtsp-server object types
11380 Fixes compiler warnings during docs build.
11381 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11383 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11386 configure: Add warning flags for compiler when configuring
11387 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11389 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11392 Automatic update of common submodule
11393 From 03a0e57 to 98e386f
11395 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11398 Automatic update of common submodule
11399 From 1fab359 to 03a0e57
11401 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11403 * gst/rtsp-server/rtsp-client.c:
11404 client: fix GSocketAddress leak in gst_rtsp_client_accept
11405 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11407 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11410 Automatic update of common submodule
11411 From f1b5a96 to 1fab359
11413 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11416 Automatic update of common submodule
11417 From 92b7266 to f1b5a96
11419 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11422 Automatic update of common submodule
11423 From ec1c4a8 to 92b7266
11425 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11428 Automatic update of common submodule
11429 From 3429ba6 to ec1c4a8
11431 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11433 * gst/rtsp-server/rtsp-auth.c:
11434 * gst/rtsp-server/rtsp-client.c:
11435 * gst/rtsp-server/rtsp-media-factory-uri.c:
11436 * gst/rtsp-server/rtsp-server.c:
11437 rtsp: fix compiler warnings
11438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11440 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11443 Automatic update of common submodule
11444 From dc70203 to 3429ba6
11446 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11448 * gst/rtsp-server/rtsp-client.c:
11449 * gst/rtsp-server/rtsp-media-factory.c:
11450 * gst/rtsp-server/rtsp-media-factory.h:
11451 * gst/rtsp-server/rtsp-media.c:
11452 * gst/rtsp-server/rtsp-media.h:
11453 * gst/rtsp-server/rtsp-server.c:
11454 * gst/rtsp-server/rtsp-server.h:
11455 * gst/rtsp-server/rtsp-session-pool.c:
11456 * gst/rtsp-server/rtsp-session-pool.h:
11457 rtsp-server: port to new thread API
11459 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11462 Automatic update of common submodule
11463 From 6db25be to dc70203
11465 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11467 * gst/rtsp-server/rtsp-auth.c:
11468 * gst/rtsp-server/rtsp-auth.h:
11469 * gst/rtsp-server/rtsp-client.c:
11470 rtsp-server: Fix compilation and compiler warnings
11472 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11476 * gst/rtsp-server/Makefile.am:
11477 configure: Modernize autotools setup a bit
11478 Also we now only create tar.bz2 and tar.xz tarballs.
11480 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11483 Automatic update of common submodule
11484 From 464fe15 to 6db25be
11486 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11489 Automatic update of common submodule
11490 From 7fda524 to 464fe15
11492 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11495 * docs/libs/Makefile.am:
11496 * docs/version.entities.in:
11497 * gst-rtsp.spec.in:
11498 * gst/rtsp-server/Makefile.am:
11499 * pkgconfig/Makefile.am:
11500 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11501 * pkgconfig/gstreamer-rtsp-server.pc.in:
11502 * tests/Makefile.am:
11503 rtsp-server: Update versioning
11505 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11507 Merge remote-tracking branch 'origin/0.10'
11509 gst/rtsp-server/rtsp-session-pool.c
11511 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11513 * gst/rtsp-server/rtsp-session-pool.c:
11514 rtsp-server: Don't use deprecated GLib API
11516 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11518 Replace master with 0.11
11520 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11522 Merge branch 'master' into 0.11
11524 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11526 Merge branch 'master' into 0.11
11528 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11531 A couple minor typo fixes
11533 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11535 * gst/rtsp-server/rtsp-media.c:
11536 media: fix state of the appqueue
11538 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11540 * gst/rtsp-server/rtsp-media-factory-uri.c:
11541 factory: use videoconvert
11543 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11545 * gst/rtsp-server/rtsp-media-factory-uri.c:
11546 factory: change to new style caps
11548 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11550 * gst/rtsp-server/rtsp-client.c:
11551 * gst/rtsp-server/rtsp-client.h:
11552 * gst/rtsp-server/rtsp-media-factory-uri.c:
11553 * gst/rtsp-server/rtsp-media.c:
11554 * gst/rtsp-server/rtsp-server.c:
11555 * gst/rtsp-server/rtsp-server.h:
11556 * gst/rtsp-server/rtsp-session-pool.c:
11557 rtsp-server: port to GIO
11560 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11563 configure: fix build
11565 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11568 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11569 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11571 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11574 * examples/Makefile.am:
11575 First rule of gst-rtsp-server club: don't talk about gst-phonon
11577 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11580 * pkgconfig/Makefile.am:
11581 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11582 * pkgconfig/gstreamer-rtsp-server.pc.in:
11583 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11584 For consistency with all other modules.
11586 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11588 * gst/rtsp-server/rtsp-client.c:
11589 rtsp-client: update for new map API
11591 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11594 * bindings/Makefile.am:
11595 * bindings/python/Makefile.am:
11596 * bindings/python/arg-types.py:
11597 * bindings/python/codegen/Makefile.am:
11598 * bindings/python/codegen/__init__.py:
11599 * bindings/python/codegen/argtypes.py:
11600 * bindings/python/codegen/code-coverage.py:
11601 * bindings/python/codegen/codegen.py:
11602 * bindings/python/codegen/definitions.py:
11603 * bindings/python/codegen/defsparser.py:
11604 * bindings/python/codegen/docextract.py:
11605 * bindings/python/codegen/docgen.py:
11606 * bindings/python/codegen/fileprefix.override:
11607 * bindings/python/codegen/fileprefixmodule.c:
11608 * bindings/python/codegen/h2def.py:
11609 * bindings/python/codegen/mergedefs.py:
11610 * bindings/python/codegen/mkskel.py:
11611 * bindings/python/codegen/override.py:
11612 * bindings/python/codegen/reversewrapper.py:
11613 * bindings/python/codegen/scmexpr.py:
11614 * bindings/python/rtspserver-types.defs:
11615 * bindings/python/rtspserver.defs:
11616 * bindings/python/rtspserver.override:
11617 * bindings/python/rtspservermodule.c:
11618 * bindings/python/test.py:
11620 python: remove pygst-based python bindings
11621 pygi is the future, apparently.
11623 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
11626 Automatic update of common submodule
11627 From c463bc0 to 7fda524
11629 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11632 Automatic update of common submodule
11633 From 2a59016 to c463bc0
11635 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11638 Automatic update of common submodule
11639 From 0807187 to 2a59016
11641 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11644 Automatic update of common submodule
11645 From 11f0cd5 to 0807187
11647 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11649 * examples/test-auth.c:
11650 example: update for new caps
11652 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11654 * examples/test-video.c:
11655 * gst/rtsp-server/rtsp-client.c:
11656 * gst/rtsp-server/rtsp-media-factory-uri.c:
11657 * gst/rtsp-server/rtsp-media.c:
11658 * gst/rtsp-server/rtsp-media.h:
11659 * gst/rtsp-server/rtsp-session.c:
11660 * gst/rtsp-server/rtsp-session.h:
11661 rtsp-server: port some more to 0.11
11663 Remove bufferlist stuff
11664 Update for new API.
11665 Add queue before appsink now that preroll-queue-len is gone.
11666 Update for request pad changes.
11668 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11670 Merge branch 'master' into 0.11
11672 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11674 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11675 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11676 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11678 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11680 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11681 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11682 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11684 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11686 Merge branch 'master' into 0.11
11688 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11690 * gst/rtsp-server/rtsp-media.c:
11691 * gst/rtsp-server/rtsp-media.h:
11692 media: add a seekable boolean
11693 Maintain the seekable state with a new variable instead of reusing the
11696 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
11698 * gst/rtsp-server/rtsp-media.c:
11699 Disallow seek in live media
11701 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11703 Merge branch 'master' into 0.11
11705 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
11707 * gst/rtsp-server/rtsp-server.c:
11708 #ifdef statements for windows socket creation were missing
11710 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
11713 Automatic update of common submodule
11714 From a39eb83 to 11f0cd5
11716 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
11719 Automatic update of common submodule
11720 From 605cd9a to a39eb83
11722 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11724 Merge branch 'master' into 0.11
11726 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11728 * gst/rtsp-server/rtsp-client.c:
11729 client: use method to access property
11731 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11733 * gst/rtsp-server/rtsp-media-factory.c:
11734 * gst/rtsp-server/rtsp-media-factory.h:
11735 media-factory: add protocols property
11736 Add a property to configure the allowed protocols in the media created from the
11739 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11741 * gst/rtsp-server/rtsp-media-factory.c:
11742 * gst/rtsp-server/rtsp-media-factory.h:
11743 media-factory: add media-configure signal
11744 Add signal to allow the application to configure the media after it was created
11747 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11749 * gst/rtsp-server/rtsp-client.c:
11750 client: use method to access property
11752 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11754 * gst/rtsp-server/rtsp-media-factory.c:
11755 * gst/rtsp-server/rtsp-media-factory.h:
11756 media-factory: add protocols property
11757 Add a property to configure the allowed protocols in the media created from the
11760 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11762 * gst/rtsp-server/rtsp-media-factory.c:
11763 * gst/rtsp-server/rtsp-media-factory.h:
11764 media-factory: add media-configure signal
11765 Add signal to allow the application to configure the media after it was created
11768 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11770 Merge branch 'master' into 0.11
11772 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11774 * gst/rtsp-server/rtsp-client.c:
11775 client: use media multicast group
11777 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11779 * gst/rtsp-server/rtsp-media-factory.h:
11780 * gst/rtsp-server/rtsp-server.h:
11781 * gst/rtsp-server/rtsp-session-pool.h:
11782 * gst/rtsp-server/rtsp-session.h:
11785 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11787 * gst/rtsp-server/rtsp-client.c:
11788 * gst/rtsp-server/rtsp-sdp.h:
11789 sdp: copy and free the server ip address
11790 Copy and free the server ip address to make memory management easier later.
11792 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11794 * gst/rtsp-server/rtsp-media-factory.c:
11795 media-factory: configure multicast in media
11797 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11799 * gst/rtsp-server/rtsp-media.c:
11800 * gst/rtsp-server/rtsp-media.h:
11801 media: add property for multicast group
11802 Add a property to configure the multicast group in the media.
11803 Based on patches from Marc Leeman and Robert Krakora.
11805 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11807 * gst/rtsp-server/rtsp-media-factory.c:
11808 * gst/rtsp-server/rtsp-media-factory.h:
11809 media-factory: add property for multicast group
11810 Add a property to configure the multicast group in the media factory.
11811 Based on patches from Marc Leeman and Robert Krakora.
11813 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11815 * gst/rtsp-server/rtsp-client.c:
11816 client: do configuration of transport in one place
11817 Move the configuration of the transport destination address to where we also
11818 configure the other bits.
11820 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11822 * gst/rtsp-server/rtsp-client.c:
11823 client: use media multicast group
11825 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11827 * gst/rtsp-server/rtsp-media-factory.h:
11828 * gst/rtsp-server/rtsp-server.h:
11829 * gst/rtsp-server/rtsp-session-pool.h:
11830 * gst/rtsp-server/rtsp-session.h:
11833 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11835 * gst/rtsp-server/rtsp-client.c:
11836 * gst/rtsp-server/rtsp-sdp.h:
11837 sdp: copy and free the server ip address
11838 Copy and free the server ip address to make memory management easier later.
11840 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11842 * gst/rtsp-server/rtsp-media-factory.c:
11843 media-factory: configure multicast in media
11845 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11847 * gst/rtsp-server/rtsp-media.c:
11848 * gst/rtsp-server/rtsp-media.h:
11849 media: add property for multicast group
11850 Add a property to configure the multicast group in the media.
11851 Based on patches from Marc Leeman and Robert Krakora.
11853 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11855 * gst/rtsp-server/rtsp-media-factory.c:
11856 * gst/rtsp-server/rtsp-media-factory.h:
11857 media-factory: add property for multicast group
11858 Add a property to configure the multicast group in the media factory.
11859 Based on patches from Marc Leeman and Robert Krakora.
11861 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11863 * gst/rtsp-server/rtsp-client.c:
11864 client: do configuration of transport in one place
11865 Move the configuration of the transport destination address to where we also
11866 configure the other bits.
11868 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11870 Merge branch 'master' into 0.11
11872 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11874 * gst/rtsp-server/rtsp-client.c:
11875 client: destroy pipeline on client disconnect with no prior TEARDOWN.
11876 The problem occurs when the client abruptly closes the connection without
11877 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
11878 server is where the pipeline gets torn down. Since this handler is not called,
11879 the pipeline remains and is up and running. Subsequent clients get their own
11880 pipelines and if the do not issue TEARDOWNs then those pipelines will also
11881 remain up and running. This is a resource leak.
11883 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11885 Merge branch 'master' into 0.11
11887 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
11889 * gst/rtsp-server/rtsp-media-factory.c:
11890 * gst/rtsp-server/rtsp-media-factory.h:
11891 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
11892 For example, it can be used to retrieve source elements like appsrc, in a more
11893 convenient way than subclassing get_element.
11895 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11897 Merge branch 'master' into 0.11
11899 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
11901 * gst/rtsp-server/rtsp-server.c:
11902 rtsp-server: hold on to reference while using object
11904 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11906 * gst/rtsp-server/rtsp-media.c:
11909 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11912 configure: use unstable api
11914 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
11916 * gst/rtsp-server/rtsp-client.c:
11917 client: fix reference counting
11919 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
11921 * gst/rtsp-server/rtsp-client.c:
11922 * gst/rtsp-server/rtsp-media.c:
11923 fix compiler warnings about unused variables
11925 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
11927 * examples/test-launch.c:
11928 * examples/test-readme.c:
11929 * examples/test-uri.c:
11930 * examples/test-video.c:
11931 examples: tell rtsp uri when ready
11933 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
11936 Automatic update of common submodule
11937 From 69b981f to 605cd9a
11939 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11941 * gst/rtsp-server/rtsp-client.c:
11942 client: update for buffer API change
11944 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11946 * gst/rtsp-server/Makefile.am:
11947 Makefile.am: 0.10 => @GST_MAJORMINOR@
11949 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11951 * gst/rtsp-server/rtsp-media-factory-uri.c:
11952 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
11954 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11956 * gst/rtsp-server/.gitignore:
11957 .gitignore: 0.10 => 0.11
11959 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11961 * gst/rtsp-server/Makefile.am:
11962 Makefile.am: 0.10 => @GST_MAJORMINOR@
11964 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11966 Merge branch 'master' into 0.11
11968 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
11971 Automatic update of common submodule
11972 From 9e5bbd5 to 69b981f
11974 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
11977 Automatic update of common submodule
11978 From fd35073 to 9e5bbd5
11980 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
11983 Automatic update of common submodule
11984 From 46dfcea to fd35073
11986 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11988 * gst/rtsp-server/rtsp-media-factory-uri.c:
11989 * gst/rtsp-server/rtsp-media.c:
11990 media: port to new caps API
11992 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11994 Merge branch 'master' into 0.11
11996 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
11998 * bindings/vala/gst-rtsp-server-0.10.vapi:
11999 Updated Vala bindings.
12000 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12002 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12004 * gst/rtsp-server/rtsp-server.c:
12005 * gst/rtsp-server/rtsp-server.h:
12006 Add a signal for newly connected clients.
12007 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12009 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12011 * bindings/python/rtspserver.override:
12012 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12014 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12016 * gst/rtsp-server/Makefile.am:
12017 * gst/rtsp-server/rtsp-client.c:
12018 * gst/rtsp-server/rtsp-funnel.c:
12019 * gst/rtsp-server/rtsp-funnel.h:
12020 * gst/rtsp-server/rtsp-media.c:
12021 rtsp-server: port to 0.11
12023 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12028 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12030 Merge branch 'master' into 0.11
12035 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12038 Automatic update of common submodule
12039 From c3cafe1 to 46dfcea
12041 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12043 * bindings/python/Makefile.am:
12044 * bindings/python/rtspserver.defs:
12045 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12047 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12049 * bindings/python/arg-types.py:
12050 python bindings: add GstRTSPUrlParam
12051 Needed to implement MediaFactory virtual proxies
12053 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12055 * bindings/python/arg-types.py:
12056 python bindings: fix returning GstRTSPUrl types
12058 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12060 * bindings/python/arg-types.py:
12061 python bindings: add arg type for GstRTSPUrl
12063 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12065 * bindings/python/rtspserver.defs:
12066 python bindings: fix the definition of MediaFactory.collect_stream
12068 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12071 Automatic update of common submodule
12072 From 1ccbe09 to c3cafe1
12074 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12077 Automatic update of common submodule
12078 From 193b717 to 1ccbe09
12080 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12083 Automatic update of common submodule
12084 From b77e2bf to 193b717
12086 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12089 build: Include lcov.mak to allow test coverage report generation
12091 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12094 Automatic update of common submodule
12095 From d8814b6 to b77e2bf
12097 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12100 Automatic update of common submodule
12101 From 6aaa286 to d8814b6
12103 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12106 Automatic update of common submodule
12107 From 6aec6b9 to 6aaa286
12109 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12112 autogen: wingo signed comment
12114 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12116 * gst/rtsp-server/rtsp-session-pool.c:
12117 session: use full charset for RTSP session ID
12118 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12119 session ID more difficult.
12120 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12122 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12124 * gst/rtsp-server/Makefile.am:
12125 rtsp-server: Don't install the funnel header
12127 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12130 Automatic update of common submodule
12131 From 1de7f6a to 6aec6b9
12133 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12136 configure: require core/base 0.10.31
12137 Needed at least for gst_plugin_feature_rank_compare_func().
12139 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12142 Automatic update of common submodule
12143 From f94d739 to 1de7f6a
12145 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12147 * gst/rtsp-server/rtsp-media.c:
12148 media: remove more unused code
12150 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12152 * gst/rtsp-server/rtsp-media.c:
12153 * gst/rtsp-server/rtsp-media.h:
12154 media: remove duplicate filtering
12155 Remove the duplicate filtering code now that we have a released -good version.
12156 Give a warning instead.
12158 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12160 * gst/rtsp-server/rtsp-media-factory.c:
12161 * gst/rtsp-server/rtsp-media.c:
12162 media: fix default buffer size
12164 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12166 * gst/rtsp-server/rtsp-media-factory.c:
12167 * gst/rtsp-server/rtsp-media-factory.h:
12168 media-factory: add property to configure the buffer-size
12169 Add a property to configure the kernel UDP buffer size.
12171 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12173 * gst/rtsp-server/rtsp-media.c:
12174 * gst/rtsp-server/rtsp-media.h:
12175 media: add property to configure kernel buffer sizes
12176 Add a property to configure the kernel UDP buffer size.
12178 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12181 configure: set PYGOBJECT_REQ before using it
12182 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12184 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12186 * docs/Makefile.am:
12187 docs: recursive into sub-directories on 'make upload'
12189 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12191 * docs/libs/gst-rtsp-server-docs.sgml:
12192 * docs/version.entities.in:
12193 docs: mention full version these docs are for, not just major-minor
12195 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12198 back to development
12200 === release 0.10.8 ===
12202 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12207 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12209 * gst/rtsp-server/rtsp-server.c:
12210 rtsp-server: clarify docs a little
12212 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12214 * gst/rtsp-server/rtsp-media.c:
12215 media: init debug category before starting thread
12217 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12219 * gst/rtsp-server/rtsp-auth.c:
12220 auth: add realm to make it more spec compliant
12222 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12224 * gst/rtsp-server/rtsp-server.c:
12225 * gst/rtsp-server/rtsp-server.h:
12226 server: add locking
12228 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12230 * examples/test-video.c:
12231 example: improve example docs a little
12233 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12235 * gst/rtsp-server/rtsp-server.c:
12236 server: ensure the watch has a ref to the server
12238 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12240 * gst/rtsp-server/rtsp-server.c:
12241 server: simpify channel function
12243 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12245 * gst/rtsp-server/rtsp-server.c:
12246 * gst/rtsp-server/rtsp-server.h:
12247 server: simplify management of channel and source
12248 We don't need to keep around the channel and source objects. Let the mainloop
12249 and the source manage the source and channel respectively.
12251 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12257 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12259 * tests/.gitignore:
12260 * tests/Makefile.am:
12261 * tests/test-cleanup.c:
12262 tests: add tests directory and cleanup test
12264 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12266 * gst/rtsp-server/rtsp-media-factory-uri.c:
12267 * gst/rtsp-server/rtsp-media-factory.c:
12268 * gst/rtsp-server/rtsp-media-mapping.c:
12269 * gst/rtsp-server/rtsp-media.c:
12270 * gst/rtsp-server/rtsp-session-pool.c:
12271 * gst/rtsp-server/rtsp-session.c:
12272 server: improve debugging in various objects
12274 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12276 * gst/rtsp-server/rtsp-server.c:
12277 server: chain up to the parent finalize
12279 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12281 * bindings/python/rtspserver-types.defs:
12282 * bindings/python/rtspserver.defs:
12283 * bindings/python/rtspserver.override:
12284 * bindings/python/test.py:
12285 gst-rtsp-server: update python bindings
12287 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12289 * gst/rtsp-server/rtsp-client.c:
12290 client: use the response from the clientstate
12291 Create the response object only once and store in the client state.
12292 Make all methods use the state response,
12294 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12296 * gst/rtsp-server/rtsp-server.c:
12297 server: use signal to keep track of clients
12298 Keep track of all the clients that the server creates and remove them when they
12299 fire the 'closed' signal.
12301 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12303 * gst/rtsp-server/rtsp-client.c:
12304 * gst/rtsp-server/rtsp-client.h:
12305 client: emit signal when closing
12307 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12309 * examples/.gitignore:
12310 * examples/Makefile.am:
12311 * examples/test-auth.c:
12312 * examples/test-video.c:
12313 * gst/rtsp-server/rtsp-auth.c:
12314 * gst/rtsp-server/rtsp-auth.h:
12315 * gst/rtsp-server/rtsp-client.c:
12316 * gst/rtsp-server/rtsp-media-factory.c:
12317 * gst/rtsp-server/rtsp-media.c:
12318 * gst/rtsp-server/rtsp-media.h:
12319 * gst/rtsp-server/rtsp-session-pool.h:
12320 * gst/rtsp-server/rtsp-session.h:
12321 media: enable per factory authorisations
12322 Allow for adding a GstRTSPAuth on the factory and media level and check
12323 permissions when accessing the factory.
12324 Add hints to the auth methods for future more fine grained authorisation.
12325 Add example application for per factory authentication.
12327 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12329 * gst/rtsp-server/rtsp-auth.c:
12330 * gst/rtsp-server/rtsp-auth.h:
12331 * gst/rtsp-server/rtsp-client.c:
12332 * gst/rtsp-server/rtsp-client.h:
12333 * gst/rtsp-server/rtsp-params.c:
12334 * gst/rtsp-server/rtsp-params.h:
12335 rtsp-server: Pass ClientState structure arround
12336 Pass the collected information for the ongoing request in a GstRTSPClientState
12337 structure that we can then pass around to simplify the method arguments. This
12338 will also be handy when we implement logging functionality.
12340 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12342 * gst/rtsp-server/rtsp-media-factory.c:
12343 * gst/rtsp-server/rtsp-media-factory.h:
12344 media-factory: add methods to configure authorisation
12346 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12348 * gst/rtsp-server/rtsp-client.c:
12349 client: unref auth in finalize
12351 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12353 * gst/rtsp-server/rtsp-server.c:
12354 server: unref auth in finalize
12356 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12358 * docs/libs/gst-rtsp-server-docs.sgml:
12359 * docs/libs/gst-rtsp-server-sections.txt:
12360 * docs/libs/gst-rtsp-server.types:
12361 docs: add more docs
12363 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12365 * gst/rtsp-server/rtsp-server.c:
12366 * gst/rtsp-server/rtsp-server.h:
12367 server: separate create and accept
12368 Create separate create and accept methods so that subclasses can create custom
12370 Configure the server in the client object and prepare for keeping track of
12373 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12375 * gst/rtsp-server/rtsp-client.c:
12376 * gst/rtsp-server/rtsp-client.h:
12377 client: add support for setting the server.
12378 Add support for keeping a ref to the server that started this client
12381 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12383 * gst/rtsp-server/rtsp-auth.c:
12384 auth: fix memleak and add some docs
12385 Fix a memleak of the basic auth token.
12386 Add docs for the helper function
12388 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12390 * gst/rtsp-server/rtsp-auth.c:
12391 * gst/rtsp-server/rtsp-auth.h:
12392 * gst/rtsp-server/rtsp-client.c:
12393 client: delegate setup of auth to the manager
12394 Delegate the configuration of the authentication tokens to the manager object
12397 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12399 * examples/test-video.c:
12400 * gst/rtsp-server/Makefile.am:
12401 * gst/rtsp-server/rtsp-auth.c:
12402 * gst/rtsp-server/rtsp-auth.h:
12403 * gst/rtsp-server/rtsp-client.c:
12404 * gst/rtsp-server/rtsp-client.h:
12405 * gst/rtsp-server/rtsp-server.c:
12406 * gst/rtsp-server/rtsp-server.h:
12407 auth: add authentication object
12408 Add an object that can check the authorization of requests.
12409 Implement basic authentication.
12410 Add example authentication to test-video
12412 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12414 * gst/rtsp-server/rtsp-server.c:
12415 * gst/rtsp-server/rtsp-server.h:
12416 server: move includes back
12417 the includes are needed for sockaddr_in.
12419 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12421 * gst/rtsp-server/rtsp-client.c:
12422 * gst/rtsp-server/rtsp-client.h:
12423 * gst/rtsp-server/rtsp-server.c:
12424 * gst/rtsp-server/rtsp-server.h:
12425 rtsp: move network includes where they are needed
12427 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12429 * gst/rtsp-server/rtsp-media.h:
12430 rtsp-media.h: Minor corrections in comments.
12433 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12436 Automatic update of common submodule
12437 From e572c87 to f94d739
12439 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12443 * docs/libs/.gitignore:
12444 * examples/.gitignore:
12445 * gst/rtsp-server/.gitignore:
12448 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12450 * docs/libs/Makefile.am:
12451 docs: We don't build ps/pdf for API reference docs
12453 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12456 Automatic update of common submodule
12457 From ccbaa85 to e572c87
12459 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12462 Automatic update of common submodule
12463 From 46445ad to ccbaa85
12465 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12467 * gst/rtsp-server/Makefile.am:
12468 * gst/rtsp-server/rtsp-funnel.c:
12469 * gst/rtsp-server/rtsp-funnel.h:
12470 * gst/rtsp-server/rtsp-media.c:
12471 funnel: rename fsfunnel to rtspfunnel
12472 Rename the funnel to avoid conflicts with the farsight one.
12474 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12476 * gst/rtsp-server/Makefile.am:
12477 * gst/rtsp-server/fs-funnel.c:
12478 * gst/rtsp-server/fs-funnel.h:
12479 * gst/rtsp-server/rtsp-media.c:
12480 rtsp-media: add and use fsfunnel
12481 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12482 select-all property that we need.
12484 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12486 * gst/rtsp-server/Makefile.am:
12487 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12488 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12489 for the g-ir-compiler, rather than just assuming the env var has
12492 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12499 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12501 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12504 * gst/rtsp-server/Makefile.am:
12505 gobject-introspection: fix g-i build for uninstalled setup
12506 Requires gst-plugins-base git (> 0.10.31.2).
12508 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12510 * examples/test-uri.c:
12511 examples: add some more options and comments
12513 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12515 * gst/rtsp-server/rtsp-media-factory-uri.c:
12516 factory-uri: use right property type
12518 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12520 * gst/rtsp-server/rtsp-media-factory-uri.c:
12521 factory-uri: attempt to configure buffer-lists
12522 Attempt to configure buffer lists in the payloader for improved performance.
12524 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12526 * gst/rtsp-server/rtsp-media.c:
12527 media: attempt to configure bigger UDP buffers
12528 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12529 send buffers with high bitrate streams.
12531 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12533 * gst/rtsp-server/rtsp-client.c:
12534 client: use the socket length from getsockname
12535 Use the length returned by getsockname to perform the getnameinfo call because
12536 the size can depend on the socket type and platform.
12539 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12541 * docs/libs/gst-rtsp-server-docs.sgml:
12542 * docs/libs/gst-rtsp-server-sections.txt:
12543 docs: add uri factory to the docs
12545 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12547 * gst/rtsp-server/rtsp-client.c:
12548 * gst/rtsp-server/rtsp-media.h:
12551 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12553 * gst/rtsp-server/rtsp-client.c:
12554 * gst/rtsp-server/rtsp-media.c:
12555 * gst/rtsp-server/rtsp-media.h:
12556 * gst/rtsp-server/rtsp-session.c:
12557 * gst/rtsp-server/rtsp-session.h:
12558 rtsp-server: add support for buffer lists
12559 Add support for sending bufferlists received from appsink.
12562 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12564 * gst/rtsp-server/rtsp-client.c:
12565 * gst/rtsp-server/rtsp-media.c:
12566 * gst/rtsp-server/rtsp-media.h:
12567 * gst/rtsp-server/rtsp-sdp.c:
12568 media: make method to retrieve the play range
12569 Make a method to retrieve the playback range so that we can conditionally create
12570 a different range for the SDP and the PLAY requests.
12572 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12574 * gst/rtsp-server/rtsp-media.c:
12575 * gst/rtsp-server/rtsp-media.h:
12576 media: add signal to notify of state changes
12578 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12580 * gst/rtsp-server/rtsp-client.h:
12581 client: cleanup headers
12583 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12585 * gst/rtsp-server/rtsp-client.c:
12588 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12590 * gst/rtsp-server/rtsp-media-factory-uri.c:
12591 * gst/rtsp-server/rtsp-media-factory-uri.h:
12592 factory-uri: add support for gstpay
12593 Add an option to prefer gstpay over decoder + raw payloader.
12595 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12597 * gst/rtsp-server/rtsp-media-factory-uri.c:
12598 * gst/rtsp-server/rtsp-media-factory-uri.h:
12599 factory-uri: rework the autoplugger.
12600 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12603 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12605 * gst/rtsp-server/rtsp-media-factory-uri.c:
12606 factory-uri: use better factory filter
12607 Make better payloader filter based on autoplug rank and RTP use case.
12609 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12612 Automatic update of common submodule
12613 From 169462a to 46445ad
12615 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12617 * gst/rtsp-server/rtsp-server.c:
12618 server: set SO_REUSEADDR before bind
12619 Set the SO_REUSEADDR _before_ bind() to make it actually work.
12621 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12623 * gst/rtsp-server/rtsp-media.c:
12624 * gst/rtsp-server/rtsp-media.h:
12625 media: emit prepared signal when prepared
12626 Make a 'prepared' signal and emit it when we successfully prepared the element.
12627 This signal can be used to configure the media object after it has been prepared
12630 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
12633 Automatic update of common submodule
12634 From 011bcc8 to 169462a
12636 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
12638 python an optional dependency
12639 * configure.ac: Move up valgrind and g-i checks. Make the python
12640 dependency optional, as it was before.
12642 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12644 Merge branch 'master' into 0.11
12649 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12651 * gst/rtsp-server/rtsp-media.c:
12652 media: update range when active clients changed
12653 When we changed the number of active clients, update the current range
12654 information because we want the second client connecting to a shared resource
12655 continue from where the stream currently.
12657 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12659 * gst/rtsp-server/rtsp-media-factory-uri.c:
12660 * gst/rtsp-server/rtsp-media-factory-uri.h:
12661 factory-uri: add colorspace and fix pt
12662 Rework the way we pass data to the autoplugger.
12663 When we have raw caps, plug a converter element to make pluggin to raw
12664 payloaders more successful.
12665 Make sure all dynamically plugged payloaders have a unique payload types.
12667 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12669 * examples/Makefile.am:
12670 * examples/test-uri.c:
12671 example: add example of the uri factory
12673 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12675 * gst/rtsp-server/Makefile.am:
12676 * gst/rtsp-server/rtsp-media-factory-uri.c:
12677 * gst/rtsp-server/rtsp-media-factory-uri.h:
12678 * gst/rtsp-server/rtsp-server.h:
12679 factory-uri: add a factory to stream any URI
12680 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
12683 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12685 * gst/rtsp-server/rtsp-media.c:
12686 * gst/rtsp-server/rtsp-media.h:
12687 media: ignore spurious ASYNC_DONE messages
12688 When we are dynamically adding pads, the addition of the udpsrc elements will
12689 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
12690 the real ASYNC_DONE when everything is prerolled.
12692 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12694 * gst/rtsp-server/rtsp-media-factory.c:
12695 * gst/rtsp-server/rtsp-media-factory.h:
12696 media-factory: make lock macro
12698 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
12700 * gst/rtsp-server/rtsp-client.c:
12701 rtsp-server: Remove unused variable and dead assignment
12703 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
12705 * examples/test-launch.c:
12706 * examples/test-mp4.c:
12707 * examples/test-ogg.c:
12708 * examples/test-readme.c:
12709 * examples/test-sdp.c:
12710 * examples/test-video.c:
12711 examples: Run gst-indent
12713 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
12715 * gst/rtsp-server/rtsp-client.c:
12716 * gst/rtsp-server/rtsp-media-factory.c:
12717 * gst/rtsp-server/rtsp-media-mapping.c:
12718 * gst/rtsp-server/rtsp-media.c:
12719 * gst/rtsp-server/rtsp-params.c:
12720 * gst/rtsp-server/rtsp-sdp.c:
12721 * gst/rtsp-server/rtsp-server.c:
12722 * gst/rtsp-server/rtsp-session-pool.c:
12723 * gst/rtsp-server/rtsp-session.c:
12724 rtsp-server: Run gst-indent
12725 Since it wasn't using the upstream common previously, there was no
12726 indentation check before commiting.
12728 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
12730 * gst/rtsp-server/rtsp-media-mapping.h:
12731 * gst/rtsp-server/rtsp-media.c:
12732 * gst/rtsp-server/rtsp-media.h:
12733 * gst/rtsp-server/rtsp-sdp.c:
12734 * gst/rtsp-server/rtsp-session-pool.h:
12735 * gst/rtsp-server/rtsp-session.c:
12736 * gst/rtsp-server/rtsp-session.h:
12737 rtsp-server: Some more doc fixups
12739 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12742 Makefile: Add cruft-cleaning support
12744 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12748 * docs/Makefile.am:
12749 * docs/libs/Makefile.am:
12750 * docs/libs/gst-rtsp-server-docs.sgml:
12751 * docs/libs/gst-rtsp-server-sections.txt:
12752 * docs/libs/gst-rtsp-server.types:
12753 * docs/version.entities.in:
12754 docs: Add gtk-doc build system
12756 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12758 * gst/rtsp-server/Makefile.am:
12759 Makefile.am: Use standard GIR make behaviour
12761 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12765 autogen/configure: Bring more in sync to standard gst module behaviour
12767 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12769 * gst/rtsp-server/rtsp-media.c:
12770 media: warn and fail when gstrtpbin is not found
12772 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12775 configure: open 0.11 branch
12777 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
12781 Add common submodule
12783 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
12785 * common/ChangeLog:
12786 * common/Makefile.am:
12787 * common/c-to-xml.py:
12788 * common/check.mak:
12789 * common/coverage/coverage-report-entry.pl:
12790 * common/coverage/coverage-report.pl:
12791 * common/coverage/coverage-report.xsl:
12792 * common/coverage/lcov.mak:
12793 * common/gettext.patch:
12794 * common/glib-gen.mak:
12795 * common/gst-autogen.sh:
12796 * common/gst-xmlinspect.py:
12798 * common/gstdoc-scangobj:
12799 * common/gtk-doc-plugins.mak:
12800 * common/gtk-doc.mak:
12801 * common/m4/.gitignore:
12802 * common/m4/Makefile.am:
12803 * common/m4/README:
12804 * common/m4/as-ac-expand.m4:
12805 * common/m4/as-auto-alt.m4:
12806 * common/m4/as-compiler-flag.m4:
12807 * common/m4/as-compiler.m4:
12808 * common/m4/as-docbook.m4:
12809 * common/m4/as-libtool-tags.m4:
12810 * common/m4/as-libtool.m4:
12811 * common/m4/as-python.m4:
12812 * common/m4/as-scrub-include.m4:
12813 * common/m4/as-version.m4:
12814 * common/m4/ax_create_stdint_h.m4:
12815 * common/m4/check.m4:
12816 * common/m4/glib-gettext.m4:
12817 * common/m4/gst-arch.m4:
12818 * common/m4/gst-args.m4:
12819 * common/m4/gst-check.m4:
12820 * common/m4/gst-debuginfo.m4:
12821 * common/m4/gst-default.m4:
12822 * common/m4/gst-doc.m4:
12823 * common/m4/gst-error.m4:
12824 * common/m4/gst-feature.m4:
12825 * common/m4/gst-function.m4:
12826 * common/m4/gst-gettext.m4:
12827 * common/m4/gst-glib2.m4:
12828 * common/m4/gst-libxml2.m4:
12829 * common/m4/gst-plugindir.m4:
12830 * common/m4/gst-valgrind.m4:
12831 * common/m4/gtk-doc.m4:
12832 * common/m4/introspection.m4:
12833 * common/m4/pkg.m4:
12834 * common/mangle-tmpl.py:
12835 * common/plugins.xsl:
12837 * common/release.mak:
12838 * common/scangobj-merge.py:
12839 * common/upload.mak:
12840 common: Remove static version
12842 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
12844 * common/m4/introspection.m4:
12845 Update introspection.m4 to match usage
12847 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12851 Remove old stuff from the README
12853 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12856 back to development
12858 === release 0.10.7 ===
12860 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12865 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12867 * examples/test-ogg.c:
12868 test-ogg: remove parsers
12869 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
12870 buffers with timestamps. Using the parsers also seems to break things.
12872 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12874 * bindings/vala/gst-rtsp-server-0.10.vapi:
12875 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12876 Updated Vala bindings
12878 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12880 * common/m4/introspection.m4:
12882 * gst/rtsp-server/Makefile.am:
12883 Added initial gobject-introspection support
12885 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12887 * gst/rtsp-server/rtsp-media-factory.c:
12888 media-factory: don't use host for shared hash key
12889 When we generate the key to share made between connections, don't include the
12890 host used to connect so that we can share media even if between clients that
12891 connected with localhost and ones with the ip address.
12893 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12895 * bindings/vala/Makefile.am:
12896 build: fix distcheck
12898 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12900 * bindings/vala/gst-rtsp-server-0.10.vapi:
12901 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12902 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12903 Update Vala bindings
12905 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12907 * bindings/vala/Makefile.am:
12909 Fix configure checks and installation location for Vala bindings
12912 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12915 back to development
12917 === release 0.10.6 ===
12919 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12922 configure: release 0.10.6
12924 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12926 * gst/rtsp-server/rtsp-media.c:
12927 media: help the compiler a little
12929 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12931 * gst/rtsp-server/rtsp-media.c:
12932 * gst/rtsp-server/rtsp-media.h:
12933 * gst/rtsp-server/rtsp-session.c:
12934 media: cleanup media transport before freeing
12935 Cleanup the media transport data before freeing. In particular, remove the qdata
12936 from the rtpsource object.
12938 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12940 * gst/rtsp-server/rtsp-media-factory.c:
12941 * gst/rtsp-server/rtsp-media-factory.h:
12942 * gst/rtsp-server/rtsp-media.c:
12943 * gst/rtsp-server/rtsp-media.h:
12944 media-factory: add eos-shutdown property
12945 Add an eos-shutdown property that will send an EOS to the pipeline before
12946 shutting it down. This allows for nice cleanup in case of a muxer.
12949 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12951 * gst/rtsp-server/rtsp-media.c:
12952 * gst/rtsp-server/rtsp-media.h:
12953 media: use multiudpsink send-duplicates when we can
12954 If we have a new enough multiudpsink with the send-duplicates property, use this
12955 instead of doing our own filtering. Our custom filtering code should eventually
12956 be removed when we can depend on a released -good.
12958 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12960 * gst/rtsp-server/rtsp-media.c:
12961 media: don't leak destinations
12962 Refactor and cleanup the destinations array when the stream is destroyed.
12964 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12966 * gst/rtsp-server/rtsp-media.c:
12967 * gst/rtsp-server/rtsp-media.h:
12968 media: don't add udp addresses multiple times
12969 Keep track of the udp addresses we added to udpsink and never add the same udp
12970 destination twice. This avoids duplicate packets when using multicast.
12972 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12974 * gst/rtsp-server/rtsp-server.c:
12975 server: disable use of SO_LINGER
12976 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
12977 server close()s the connection.
12979 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12981 * gst/rtsp-server/rtsp-server.c:
12982 server: use 5 second linger period in SO_LINGER
12983 Wait 5 seconds before clearing the send buffers and reseting the connection with
12984 the client when we do a close. This should be enough time to get the message to
12988 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12990 * gst/rtsp-server/rtsp-server.c:
12991 server: use SO_LINGER
12992 SO_LINGER on the socket will make sure that any pending data on the socket is
12993 flushed ASAP and that the socket connection is reset. This makes sure that the
12994 socket can be reused immediately.
12997 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13000 README: add blurb about shared media factories
13002 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13004 * gst/rtsp-server/rtsp-media.c:
13005 Add stdlib.h for atoi()
13007 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13009 * bindings/python/Makefile.am:
13010 * bindings/vala/Makefile.am:
13011 build: distcheck fixes
13012 Fix 'make distcheck', somewhat (it still fails because it tries to
13013 install files into /usr/share/vala/vapi/ irrespective of the
13014 configured prefix).
13016 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13019 configure: bump core/base requirements to released version
13020 Makes things less confusing for people.
13022 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13025 configure: fail if GStreamer core/base requirements are not met
13027 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13029 * gst/rtsp-server/rtsp-client.c:
13030 client: improve client cleanups
13031 Make sure the session does not timeout when using TCP. We need to do this
13032 because quicktime player does not send RTCP for some reason in tunneled
13034 Refactor some cleanup code.
13037 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13039 * gst/rtsp-server/rtsp-session.c:
13040 * gst/rtsp-server/rtsp-session.h:
13041 session: add support for prevent session timeouts
13042 Add an atomix counter to prevent session timeouts when we are, for example,
13043 streaming over TCP.
13045 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13047 * gst/rtsp-server/rtsp-client.c:
13048 client: fix unlink on session timeouts
13049 When our session times out, make sure we unlink all streams in this
13051 Remove the tunnelid when closing the connection.
13053 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13055 * gst/rtsp-server/rtsp-session.c:
13056 session: small cleanups
13058 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13060 * gst/rtsp-server/rtsp-client.c:
13061 client: handle lost_tunnel callbacks
13062 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13063 hashtable so that we can reuse it for when the client reopens the POST
13065 Close the connection after a TEARDOWN.
13066 Make sure or watchid is cleared when the watch is removed.
13069 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13071 * gst/rtsp-server/rtsp-client.c:
13072 * gst/rtsp-server/rtsp-media.c:
13073 * gst/rtsp-server/rtsp-sdp.c:
13074 rtsp-server: add more support for multicast
13076 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13079 * gst/rtsp-server/rtsp-media.c:
13080 * gst/rtsp-server/rtsp-media.h:
13081 media: allow configuration of allowed lower transport
13083 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13085 * gst/rtsp-server/rtsp-client.h:
13086 * gst/rtsp-server/rtsp-media.c:
13087 * gst/rtsp-server/rtsp-media.h:
13088 * gst/rtsp-server/rtsp-sdp.c:
13089 * gst/rtsp-server/rtsp-sdp.h:
13090 * gst/rtsp-server/rtsp-server.c:
13091 rtsp: keep track of server ip and ipv6
13092 Keep track of how the client connected to the server and setup the udp ports
13093 with the same protocol.
13094 Copy the server ip address in the SDP so that clients can send RTCP back to
13097 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13099 * gst/rtsp-server/rtsp-session.c:
13102 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13104 * gst/rtsp-server/rtsp-client.c:
13105 client: use right size for malloc
13107 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13109 * gst/rtsp-server/rtsp-server.c:
13110 server: comment ipv6 server listening address
13112 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13114 * gst/rtsp-server/rtsp-media.c:
13115 media: allow for ipv6 sockets
13117 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13119 * gst/rtsp-server/rtsp-server.c:
13120 * gst/rtsp-server/rtsp-server.h:
13121 server: rework server part
13122 Allow setting a bind address, make sure we can deal with ipv6.
13123 Remove the port property and change with the service property.
13125 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13127 * gst/rtsp-server/rtsp-media.h:
13128 media: update comments a little
13130 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13132 * gst/rtsp-server/rtsp-client.c:
13133 client: make content-base better
13134 Use the URI formatting functions to make a content-base. Also make sure that
13135 there is a trailing / at the end.
13137 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13139 * gst/rtsp-server/rtsp-client.c:
13140 client: guard against invalid paths
13142 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13144 * examples/test-video.c:
13145 test: catch server bind errors
13147 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13149 * gst/rtsp-server/rtsp-media.c:
13150 rtspmedia: emit "unprepared" if _prepare fails.
13151 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13152 media object is removed from its factory's cache.
13154 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13156 * gst/rtsp-server/rtsp-media.c:
13157 media: collect media position when seek completes
13159 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13161 * gst/rtsp-server/rtsp-client.c:
13162 client: call unlink_streams in client finalize
13165 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13167 * gst/rtsp-server/rtsp-media.c:
13168 media: limit the time to wait to something huge
13169 Avoid waiting forever but limit the timeout to 20 seconds.
13171 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13173 * gst/rtsp-server/rtsp-sdp.c:
13174 sdp: reindent and check for prepared status
13176 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13178 * gst/rtsp-server/rtsp-media.c:
13179 * gst/rtsp-server/rtsp-media.h:
13180 * gst/rtsp-server/rtsp-session.c:
13181 media: avoid doing _get_state() for state changes
13182 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13183 until the media is prerolled or in error. This avoids doing a blocking call of
13184 gst_element_get_state() that can cause lockups when there is an error.
13187 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13189 * gst/rtsp-server/rtsp-media.c:
13192 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13194 * gst/rtsp-server/rtsp-media-factory.c:
13195 media-factory: better error handling
13196 Improve the error handling a bit.
13198 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13200 * gst/rtsp-server/rtsp-client.c:
13201 client: rework transport parsing
13202 Rework the transport parsing code so that we can ignore transports we don't
13203 support instead of just picking the first one we can parse.
13204 Configure a (for now hardcoded) destination for multicast transports.
13206 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13208 * gst/rtsp-server/rtsp-media.c:
13209 media: set multicast sink parameters
13210 Disable loop and automatic multicast join on the udpsink elements.
13211 Add some more debug info.
13212 Reset some state variables in the right place.
13213 Use the right port numbers for multicast.
13215 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13217 * gst/rtsp-server/rtsp-session.c:
13218 session: handle transport setup correctly
13219 Handle UDP, MCAST and TCP transport negotiation more correctly.
13220 Store the server session SSRC in the transport.
13222 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13224 * gst/rtsp-server/rtsp-client.c:
13225 rtsp-client: implement error_full
13226 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13229 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13232 * gst/rtsp-server/rtsp-client.c:
13233 * gst/rtsp-server/rtsp-server.c:
13234 docs: update docs and comments
13236 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13238 * gst/rtsp-server/rtsp-sdp.c:
13239 sdp: make server work better when behind a proxy
13241 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13243 * gst/rtsp-server/rtsp-client.c:
13244 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13246 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13248 * gst/rtsp-server/rtsp-client.c:
13249 * gst/rtsp-server/rtsp-media-factory.c:
13250 * gst/rtsp-server/rtsp-media-mapping.c:
13251 * gst/rtsp-server/rtsp-media.c:
13252 * gst/rtsp-server/rtsp-server.c:
13253 * gst/rtsp-server/rtsp-session-pool.c:
13254 * gst/rtsp-server/rtsp-session.c:
13255 Use GStreamer's debugging subsystem
13257 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13259 * gst/rtsp-server/rtsp-media-factory.c:
13260 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13262 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13265 back to development
13267 === release 0.10.5 ===
13269 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13274 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13277 configure: bump required versions
13279 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13281 * gst/rtsp-server/rtsp-client.c:
13282 client: call weak-unref on client->sessions from finalize
13285 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13287 * gst/rtsp-server/rtsp-media.c:
13288 media: Fixed crasher where caps got unref'ed too often
13290 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13293 * pkgconfig/.gitignore:
13294 * pkgconfig/Makefile.am:
13295 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13296 Added pkg-config file to use gst-rtsp-server uninstalled
13298 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13300 * gst/rtsp-server/rtsp-media.c:
13301 media: add some docs
13303 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13305 * gst/rtsp-server/rtsp-client.c:
13306 rtsp: Use gst_rtsp_watch_send_message().
13307 Use gst_rtsp_watch_send_message() since the old API which used
13308 gst_rtsp_watch_queue_message() has been deprecated.
13310 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13313 back to development
13315 === release 0.10.4 ===
13317 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13322 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13324 * gst/rtsp-server/rtsp-client.c:
13325 * gst/rtsp-server/rtsp-session.c:
13326 * gst/rtsp-server/rtsp-session.h:
13327 rtsp: allocate channels in TCP mode
13328 When the client does not provide us with channels in TCP mode, allocate channels
13331 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13333 * gst/rtsp-server/rtsp-client.c:
13334 client: don't crash when tunnelid is missing
13335 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13336 don't crash but return an error response to the client.
13339 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13341 * bindings/vala/gst-rtsp-server-0.10.vapi:
13342 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13343 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13344 bindings: update vala bindings with new method
13346 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13348 * gst/rtsp-server/rtsp-session-pool.c:
13349 * gst/rtsp-server/rtsp-session-pool.h:
13350 sessionpool: add function to filter sessions
13351 Add generic function to retrieve/remove sessions.
13353 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13356 configure: bump core/base requirements to release
13358 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13360 * gst/rtsp-server/rtsp-media.c:
13361 media: fix indentation
13363 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13365 * gst/rtsp-server/rtsp-media.c:
13366 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13368 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13370 * gst/rtsp-server/rtsp-media.c:
13371 set state and remove elements of media in for loop
13373 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13375 * bindings/vala/gst-rtsp-server-0.10.vapi:
13376 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13377 Added gst_rtsp_media_remove_elements function to Vala bindings
13379 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13381 * gst/rtsp-server/rtsp-media.c:
13382 * gst/rtsp-server/rtsp-media.h:
13383 Added gst_rtsp_media_remove_elements function
13385 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13387 * gst/rtsp-server/rtsp-media.c:
13388 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13390 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13392 * bindings/vala/gst-rtsp-server-0.10.vapi:
13393 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13394 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13395 Updated Vala bindings
13397 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13399 * gst/rtsp-server/rtsp-media.c:
13400 * gst/rtsp-server/rtsp-media.h:
13401 Added vmethod unprepare to GstRTSPMedia
13402 The default implementation sets the state of the pipeline to GST_STATE_NULL
13404 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13406 * gst/rtsp-server/rtsp-media-factory.c:
13407 * gst/rtsp-server/rtsp-media-factory.h:
13408 Made collect_streams function public
13410 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13412 * gst/rtsp-server/rtsp-media-factory.c:
13413 * gst/rtsp-server/rtsp-media-factory.h:
13414 * gst/rtsp-server/rtsp-media.c:
13415 Added vmethod create_pipeline to GstRTSPMediaFactory
13416 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13418 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13420 * gst/rtsp-server/rtsp-client.c:
13421 client: use g_source_destroy()
13422 We need to use g_source_destroy() because we might have added the source to a
13423 different main context than the default one.
13425 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13427 * gst/rtsp-server/Makefile.am:
13428 * gst/rtsp-server/rtsp-client.c:
13429 * gst/rtsp-server/rtsp-params.c:
13430 * gst/rtsp-server/rtsp-params.h:
13431 rtsp: prepare for handling GET/SET_PARAMETER
13432 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13434 Fix return codes of handlers.
13436 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13438 * gst/rtsp-server/rtsp-media.c:
13439 media: don't leak session pads
13441 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13443 * gst/rtsp-server/rtsp-media.c:
13444 media: clean up the messages a bit
13446 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13448 * gst/rtsp-server/rtsp-sdp.c:
13449 sdp: warn and skip streams without media
13451 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13453 * bindings/vala/gst-rtsp-server-0.10.vapi:
13454 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13455 vala: Fixed typo in header file of RTSPMediaStream
13457 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13459 * gst/rtsp-server/rtsp-media.c:
13461 Fix a debug message
13462 Make dumping RTCP stats configurable
13464 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13466 * gst/rtsp-server/rtsp-media.c:
13467 media: be less verbose and leak less
13469 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13471 * gst/rtsp-server/rtsp-media.c:
13472 media: don't leak the destination address
13474 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13476 * gst/rtsp-server/rtsp-client.c:
13477 * gst/rtsp-server/rtsp-media.c:
13478 * gst/rtsp-server/rtsp-media.h:
13479 * gst/rtsp-server/rtsp-session.c:
13480 * gst/rtsp-server/rtsp-session.h:
13481 rtsp: use RTCP to keep the session alive
13482 Use the RTCP rtcp-from stats field to find the associated session and use this
13483 to keep the session alive.
13485 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13487 * gst/rtsp-server/rtsp-session.c:
13488 session: add 5sec to the real session timeout
13489 Allow the session to live 5sec longer before really timing out. This should give
13490 clients some extra time to keep the session active.
13492 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13494 * gst/rtsp-server/rtsp-client.c:
13495 client: replay OK to GET/SET_PARAMETER
13496 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13497 so that we return OK for those requests.
13499 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13501 * gst/rtsp-server/rtsp-media.c:
13502 * gst/rtsp-server/rtsp-media.h:
13503 media: keep track of active transports
13504 Keep track of which transport is active to avoid closing the connection too
13506 Remove the destination transport also when going to NULL.
13507 Print some stats about the SDES and other RTCP messages we receive from the
13510 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13512 * examples/.gitignore:
13513 * examples/Makefile.am:
13514 * examples/test-sdp.c:
13515 example: add SDP relay example
13517 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13519 * gst/rtsp-server/rtsp-media.c:
13520 media: also count active TCP connections
13522 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13524 * gst/rtsp-server/rtsp-media-factory.c:
13525 * gst/rtsp-server/rtsp-media.c:
13526 * gst/rtsp-server/rtsp-media.h:
13527 rtsp: add support for dynamic elements
13528 Add support for dynamic elements.
13529 Don't set live pipelines back to paused.
13531 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13533 * gst/rtsp-server/rtsp-sdp.c:
13534 sdp: don't add encoding name when absent in caps
13536 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13538 * gst/rtsp-server/rtsp-client.c:
13539 client: warn when we can't do RTP-Info
13541 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13543 * gst/rtsp-server/rtsp-media-factory.c:
13544 factory: factor out the stream construction
13546 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13548 * gst/rtsp-server/rtsp-client.c:
13549 client: only add RTP-Info when we have the info
13550 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13553 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13556 back to development
13558 === release 0.10.3 ===
13560 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13564 - Fixes a bug where it put the wrong verion in pkgconfig
13565 - Link RTP and RTCP sources
13567 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13569 * gst/rtsp-server/rtsp-media.c:
13570 * gst/rtsp-server/rtsp-media.h:
13571 media: link the RTP udpsrc to the session manager
13572 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13573 shut down when the client sends a packet to open firewalls.
13575 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13577 * pkgconfig/gst-rtsp-server.pc.in:
13578 Don't use hard-coded version number in pkg-config file
13580 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13583 back to development
13585 === release 0.10.2 ===
13587 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13592 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13595 * common/m4/.gitignore:
13596 * examples/.gitignore:
13597 * pkgconfig/.gitignore:
13598 add some .gitignore files
13600 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13602 * gst/rtsp-server/rtsp-media.c:
13603 media: seek to key frames
13605 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13607 * gst/rtsp-server/rtsp-media.c:
13608 media: emit the unprepared signal by id
13609 Emit the unprepared signal by id instead of name and set the media as
13612 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13614 * gst/rtsp-server/rtsp-media.c:
13615 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13617 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13619 * gst/rtsp-server/rtsp-server.c:
13620 Added finalize function to GstRTPSPServer to unref session pool and media mapping
13622 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13624 * bindings/vala/gst-rtsp-server-0.10.vapi:
13625 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13626 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13627 Updated vala bindings
13629 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13631 * gst/rtsp-server/Makefile.am:
13632 * gst/rtsp-server/rtsp-client.c:
13633 * gst/rtsp-server/rtsp-media.c:
13634 server: use appsink and appsrc with the API
13635 Use the appsink/appsrc API instead of the signals for higher
13638 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13640 * examples/test-ogg.c:
13641 tests: set the payload type correctly
13643 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13645 * gst/rtsp-server/rtsp-media-factory.c:
13646 factory: connect to the unprepare signal
13647 Connect to the unprepare signal for non-reusable media so that we can remove
13648 them from the cache.
13650 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13652 * gst/rtsp-server/rtsp-media.c:
13653 * gst/rtsp-server/rtsp-media.h:
13654 media: add signal to notify of unprepare
13656 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13658 * gst/rtsp-server/rtsp-media.c:
13659 * gst/rtsp-server/rtsp-media.h:
13660 media: more work on making the media shared
13661 Add a reusable flag to medias, indicating that they can be reused after a state
13665 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13667 * examples/test-readme.c:
13668 examples: mark the example as shared for testing
13670 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13672 * gst/rtsp-server/rtsp-media.c:
13673 * gst/rtsp-server/rtsp-media.h:
13674 client: support shared media
13675 Always perform the state actions even if the target state of the pipeline is
13676 already correct, we still want to add/remove the transports when we are dealing
13678 Keep a counter of the number of active transports for a media so that we can use
13679 this to perform a state change when needed.
13680 Perform a state change of the pipeline only when the first transport was added
13681 or when there are no active transports.
13683 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13685 * gst/rtsp-server/rtsp-client.c:
13686 client: fix refcounting crasher
13687 Don't need to remove the weak refs in the finalize methods, they are already
13688 removed in the dispose.
13689 Don't register the callback with a DestroyNofity.
13691 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13693 * gst/rtsp-server/rtsp-client.c:
13694 Fix rtsp client refcount management in TCP mode.
13695 Don't unref a client ref we never had. Fixes an unref
13696 of an already-free client object after a client
13697 teardown request for me.
13699 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13701 * gst/rtsp-server/rtsp-session.c:
13702 docs: fix typo in API docs
13704 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13706 * gst/rtsp-server/rtsp-media.c:
13707 More seeking fixes.
13708 Keep the udp sources in playing even if we go to paused. unlock the sources when
13710 Add some more debug info.
13711 Only seek when we need to.
13712 Keep track of the position when we go to paused.
13714 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13716 * gst/rtsp-server/rtsp-client.c:
13717 * gst/rtsp-server/rtsp-media.c:
13718 * gst/rtsp-server/rtsp-media.h:
13719 Add beginnings of seeking.
13720 Parse the Range header and perform a seek on the pipeline for the requested
13721 position. It's disabled currently until I figure out what's going wrong.
13723 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13725 * gst/rtsp-server/rtsp-client.c:
13726 allow pause requests for now.
13729 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13731 * gst/rtsp-server/rtsp-client.c:
13732 Remove weak ref on the session in teardown
13733 We need to remove our weakref from the session when we do a teardown because
13734 else we close the TCP connection prematurely.
13736 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13738 * gst/rtsp-server/rtsp-client.c:
13739 * gst/rtsp-server/rtsp-client.h:
13740 * gst/rtsp-server/rtsp-session-pool.c:
13741 Do some more session cleanup
13742 Make session timeout kill the TCP connection that currently watches the
13744 Remove the client timeout property.
13746 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13748 * gst/rtsp-server/rtsp-client.c:
13749 * gst/rtsp-server/rtsp-client.h:
13750 * gst/rtsp-server/rtsp-media.c:
13751 * gst/rtsp-server/rtsp-media.h:
13752 * gst/rtsp-server/rtsp-server.c:
13753 * gst/rtsp-server/rtsp-session.c:
13754 * gst/rtsp-server/rtsp-session.h:
13756 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
13759 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13761 * examples/Makefile.am:
13762 * examples/test-launch.c:
13763 Add example server that takes launch lines
13764 Add an example server that streams any -launch line.
13766 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13768 * examples/test-readme.c:
13769 * gst/rtsp-server/rtsp-client.c:
13770 * gst/rtsp-server/rtsp-media.c:
13771 * gst/rtsp-server/rtsp-media.h:
13772 Add support for live streams
13773 Add support for live streams and ranges
13774 Start on handling TCP data transfer.
13776 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13778 * gst/rtsp-server/rtsp-media.c:
13779 Free the pipeline before other things
13782 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13784 * gst/rtsp-server/rtsp-client.c:
13785 Only free the pending tunnel if there is one
13788 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13790 * gst/rtsp-server/rtsp-client.c:
13791 * gst/rtsp-server/rtsp-client.h:
13792 * gst/rtsp-server/rtsp-media.c:
13793 rtsp-server: Add support for tunneling
13794 Add support for tunneling over HTTP.
13795 Use new connection methods to retrieve the url.
13796 Dispatch messages based on the message type instead of blindly
13797 assuming it's always a request.
13798 Keep track of the watch id so that we can remove it later.
13799 Set the media pipeline to NULL before unreffing the pipeline.
13801 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13803 * gst/rtsp-server/rtsp-client.c:
13804 * gst/rtsp-server/rtsp-client.h:
13805 Fix for channel -> watch rename in gstreamer
13806 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
13808 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13810 * gst/rtsp-server/rtsp-client.c:
13811 * gst/rtsp-server/rtsp-client.h:
13813 Use the async RTSP channels instead of spawning a new thread for each client.
13814 If a sessionid is specified in a request, fail if we don't have the session.
13816 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13818 * gst/rtsp-server/rtsp-media.c:
13819 Add better debug info
13820 Add some better debug info.
13822 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13824 * examples/test-video.c:
13826 Add support for session timeouts in the example.
13828 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13830 * gst/rtsp-server/rtsp-session-pool.c:
13831 * gst/rtsp-server/rtsp-session-pool.h:
13832 Pass GTimeVal around for performance reasons
13833 Get the current time only once and pass it around so that sessions don't have to
13834 get the current time anymore.
13835 Add experimental support for a GSource that dispatches when the session needs to
13838 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13840 * gst/rtsp-server/rtsp-session.c:
13841 * gst/rtsp-server/rtsp-session.h:
13842 Add better support for session timeouts
13843 Add a method to request the number of milliseconds when a session will timeout.
13845 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13847 * gst/rtsp-server/rtsp-media.c:
13848 * gst/rtsp-server/rtsp-media.h:
13849 Add suport for RTP manager monitoring
13850 Add the first stage in monitoring the rtp manager.
13851 Make sure we don't update the state to something we don't want.
13853 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13855 * gst/rtsp-server/rtsp-client.c:
13856 Add support for session keepalive
13857 Get and update the session timeout for all requests. get the session as early as
13860 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13862 * gst/rtsp-server/rtsp-media-factory.h:
13863 * gst/rtsp-server/rtsp-media.c:
13864 * gst/rtsp-server/rtsp-media.h:
13865 Handle media bus messages
13866 Handle media bus messages in a custom mainloop and dispatch them to the
13867 RTSPMedia objects. Let the default implementation handle some common messages.
13869 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13871 * gst/rtsp-server/rtsp-client.c:
13872 * gst/rtsp-server/rtsp-session-pool.c:
13873 * gst/rtsp-server/rtsp-session.c:
13874 Some more session timeout handling
13875 Move the session header setting code to a central place so that we always add
13876 the timeout parameter too.
13877 Handle timeouts by running the session cleanup code.
13878 Stop media before cleaning up.
13880 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13882 * gst/rtsp-server/rtsp-client.c:
13883 * gst/rtsp-server/rtsp-client.h:
13884 Add timeout property
13885 Add a timeout property ot the client and make the other properties into GObject
13888 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13890 * gst/rtsp-server/rtsp-session-pool.c:
13891 Use getters and setters in property code
13892 Use the getters and setters for the timeout property instead of locking
13895 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13897 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
13899 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13901 * gst/rtsp-server/rtsp-session-pool.c:
13902 * gst/rtsp-server/rtsp-session-pool.h:
13903 * gst/rtsp-server/rtsp-session.c:
13904 * gst/rtsp-server/rtsp-session.h:
13905 Add more timeout stuff
13906 Add method to check if a session is expired.
13907 Add method to perform cleanup on a session pool.
13909 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13911 * gst/rtsp-server/rtsp-client.c:
13912 * gst/rtsp-server/rtsp-session-pool.c:
13913 * gst/rtsp-server/rtsp-session-pool.h:
13914 * gst/rtsp-server/rtsp-session.c:
13915 * gst/rtsp-server/rtsp-session.h:
13916 Add beginnings of session timeouts and limits
13917 Add the timeout value to the Session header for unusual timeout values.
13918 Allow us to configure a limit to the amount of active sessions in a pool. Set a
13919 limit on the amount of retry we do after a sessionid collision.
13920 Add properties to the sessionid and the timeout of a session. Keep track of
13921 creation time and last access time for sessions.
13923 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13925 * gst/rtsp-server/rtsp-client.c:
13926 * gst/rtsp-server/rtsp-media.c:
13927 * gst/rtsp-server/rtsp-media.h:
13928 * gst/rtsp-server/rtsp-sdp.c:
13929 * gst/rtsp-server/rtsp-session-pool.c:
13930 * gst/rtsp-server/rtsp-session.c:
13931 * gst/rtsp-server/rtsp-session.h:
13932 Cleanup of sessions and more
13933 Fix the refcounting of media and sessions in the client. Properly clean up the
13934 session data when the client performs a teardown.
13935 Add Server header to responses.
13936 Allow for multiple uri setups in one session.
13937 Add Range header to the PLAY response and add the range attribute to the SDP
13939 Fix the session pool remove method, it used the wrong key in the hashtable. Also
13940 give the ownership of the sessionid to the session object.
13942 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13944 * gst/rtsp-server/rtsp-server.c:
13945 * gst/rtsp-server/rtsp-server.h:
13947 Rename the 'server_port' variable to simply 'port'.
13949 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13952 * gst/rtsp-server/rtsp-client.c:
13953 * gst/rtsp-server/rtsp-media.c:
13954 * gst/rtsp-server/rtsp-media.h:
13955 * gst/rtsp-server/rtsp-session.c:
13956 * gst/rtsp-server/rtsp-session.h:
13957 Rework the way we handle transports for streams
13958 Make the media accept an array of transports for the streams that we have
13959 configured for the play/pause requests.
13960 Implement server states for a client and its media.
13961 Require 0.10.22.1 (git HEAD) of gstreamer.
13963 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13965 * gst/rtsp-server/rtsp-client.c:
13966 * gst/rtsp-server/rtsp-media-factory.c:
13967 Drop const from functions dealing with urls
13968 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
13969 have the right const in them.
13971 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13973 * gst/rtsp-server/rtsp-client.c:
13974 * gst/rtsp-server/rtsp-media.c:
13975 * gst/rtsp-server/rtsp-sdp.c:
13979 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13981 * gst/rtsp-server/rtsp-client.c:
13982 * gst/rtsp-server/rtsp-media-factory.c:
13983 * gst/rtsp-server/rtsp-media.c:
13984 * gst/rtsp-server/rtsp-media.h:
13986 Don't keep a reference to the GstRTSPMedia in the stream.
13987 Free more things when freeing the GstRTSPMedia.
13989 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13992 * gst/rtsp-server/rtsp-media-factory.c:
13993 * gst/rtsp-server/rtsp-media-factory.h:
13994 * gst/rtsp-server/rtsp-media.c:
13995 * gst/rtsp-server/rtsp-media.h:
13996 * gst/rtsp-server/rtsp-server.c:
13997 * gst/rtsp-server/rtsp-server.h:
13998 More docs and small cleanups
13999 Add some more docs and update the README
14000 Cleanup some method names.
14001 Remove an unneeded idx field in the GstRTSPMediaStream
14003 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14006 * examples/Makefile.am:
14007 * examples/test-readme.c:
14008 Add a README and more example code
14009 Add a README file that contains a small introduction on how to use the server
14010 along with the example code explained in the readme.
14012 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14014 * gst/rtsp-server/rtsp-media.c:
14015 * gst/rtsp-server/rtsp-server.c:
14016 Fix some leaks and change default port
14017 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14018 we finished the initial preroll. If we keep them locked, setting the pipeline to
14019 NULL will not stop and clean up the sources correctly.
14020 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14022 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14024 * gst/rtsp-server/rtsp-session.c:
14025 * gst/rtsp-server/rtsp-session.h:
14026 Cleanups to the session object
14027 Remove some unneeded variables in the session state of a stream such as the
14028 owner media and the server transport.
14029 Get the configuration of a media stream in a session based on the media_stream
14030 in the original object instead of our cached index.
14031 Free more data in the finalize method.
14033 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14035 * gst/rtsp-server/rtsp-client.c:
14036 * gst/rtsp-server/rtsp-client.h:
14037 Cleanups and reuse media from DESCRIBE
14038 Handle thread create errors.
14039 Rename some internal methods to better match what they actually do.
14040 Handle misconfiguration of session_pool and media_mapping gracefully.
14041 Cache the DESCRIBE media and uri in the client connection and reuse them when
14042 we receive a SETUP request in the same connection for the same uri.
14043 Cleanup the client connection object.
14045 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14047 * gst/rtsp-server/rtsp-media-factory.c:
14048 * gst/rtsp-server/rtsp-media-factory.h:
14049 * gst/rtsp-server/rtsp-media.c:
14050 * gst/rtsp-server/rtsp-media.h:
14051 Add shared properties to media and factory
14052 Add the shared property to media.
14053 Implement some simple caching in the factory depending on if the media is shared
14056 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14058 * gst/rtsp-server/rtsp-client.c:
14059 Add a little comment
14060 Add some comment about the content-base header.
14062 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14064 * examples/Makefile.am:
14065 * examples/test-mp4.c:
14066 * examples/test-ogg.c:
14067 * examples/test-video.c:
14068 * gst/rtsp-server/Makefile.am:
14069 * gst/rtsp-server/rtsp-client.c:
14070 * gst/rtsp-server/rtsp-client.h:
14071 * gst/rtsp-server/rtsp-media-factory.c:
14072 * gst/rtsp-server/rtsp-media-factory.h:
14073 * gst/rtsp-server/rtsp-media.c:
14074 * gst/rtsp-server/rtsp-media.h:
14075 * gst/rtsp-server/rtsp-sdp.c:
14076 * gst/rtsp-server/rtsp-sdp.h:
14077 * gst/rtsp-server/rtsp-server.c:
14078 * gst/rtsp-server/rtsp-server.h:
14079 * gst/rtsp-server/rtsp-session.c:
14080 * gst/rtsp-server/rtsp-session.h:
14081 Reorganize things, prepare for media sharing
14082 Added various other test server examples
14083 Move the SDP message generation to a separate helper.
14084 Refactor common code for finding the session.
14085 Add content-base for realplayer compatibility
14086 Clean up request uris before processing for better vlc compatibility.
14087 Move prerolling and pipeline construction to the RTSPMedia object.
14088 Use multiudpsink for future pipeline reuse.
14090 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14093 Back to development
14096 === release 0.10.1 ===
14098 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14101 Make 0.10.1 release
14104 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14106 * bindings/vala/Makefile.am:
14108 Add more directories and files to the dist.
14110 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14112 * bindings/python/Makefile.am:
14113 * bindings/python/rtspserver.override:
14114 Fixed compile error of python bindings
14116 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14118 * bindings/vala/gst-rtsp-server-0.10.vapi:
14119 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14120 Marked values as nullable accordingly
14122 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14124 * bindings/vala/gst-rtsp-server-0.10.vapi:
14125 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14126 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14127 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14128 Updated Vala bindings
14130 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14132 * gst/rtsp-server/rtsp-client.c:
14133 * gst/rtsp-server/rtsp-media-mapping.c:
14134 * gst/rtsp-server/rtsp-media-mapping.h:
14135 * gst/rtsp-server/rtsp-media.h:
14136 * gst/rtsp-server/rtsp-session-pool.h:
14137 Cleanups and doc updates
14138 Add some more documentation and do some minor cleanups here and there.
14140 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14142 * gst/rtsp-server/rtsp-client.c:
14143 * gst/rtsp-server/rtsp-media-factory.c:
14144 * gst/rtsp-server/rtsp-media-factory.h:
14145 * gst/rtsp-server/rtsp-media.c:
14146 * gst/rtsp-server/rtsp-media.h:
14147 * gst/rtsp-server/rtsp-session.c:
14148 * gst/rtsp-server/rtsp-session.h:
14150 Rename GstRTSPMediaBin to GstRTSPMedia
14151 Parse the request url into a GstRTSPUri object and pass this object to the
14152 various handlers and methods that require the uri.
14154 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14158 Add some more docs and remove some old code from the example.
14160 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14162 * gst/rtsp-server/rtsp-client.c:
14163 Handle state change failures better
14164 Handle state change failures better when changing the state of the pipeline to
14167 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14169 * gst/rtsp-server/rtsp-media-factory.c:
14170 * gst/rtsp-server/rtsp-media-factory.h:
14171 Make element creation more extendible
14172 Add get_element vmethod to the default MediaFactory so that subclasses can just
14173 override that method and still use the default logic for making a MediaBin from
14176 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14179 * gst/rtsp-server/Makefile.am:
14180 * gst/rtsp-server/rtsp-client.c:
14181 * gst/rtsp-server/rtsp-client.h:
14182 * gst/rtsp-server/rtsp-media-factory.c:
14183 * gst/rtsp-server/rtsp-media-factory.h:
14184 * gst/rtsp-server/rtsp-media-mapping.c:
14185 * gst/rtsp-server/rtsp-media-mapping.h:
14186 * gst/rtsp-server/rtsp-media.c:
14187 * gst/rtsp-server/rtsp-media.h:
14188 * gst/rtsp-server/rtsp-server.c:
14189 * gst/rtsp-server/rtsp-server.h:
14190 * gst/rtsp-server/rtsp-session.c:
14191 * gst/rtsp-server/rtsp-session.h:
14192 Make the server handle arbitrary pipelines
14193 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14194 The GstMediaBin object has a handle to a bin with elements and to a list of
14195 GstMediaStream objects that this bin produces.
14196 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14197 with methods to register and remove those mappings.
14198 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14199 used by the server instance.
14200 Modify the example application so that it shows how to create custom pipelines
14201 attached to a specific mount point.
14202 Various misc cleanps.
14204 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14206 * gst/rtsp-server/rtsp-server.c:
14207 * gst/rtsp-server/rtsp-server.h:
14208 Allow setting a custom media factory for a server
14210 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14212 * gst/rtsp-server/rtsp-client.c:
14213 * gst/rtsp-server/rtsp-client.h:
14214 Allow setting a custom media factory for a client.
14216 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14218 * gst/rtsp-server/Makefile.am:
14219 Add Makefile entry for the media factory
14221 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14223 * gst/rtsp-server/rtsp-media-factory.c:
14224 * gst/rtsp-server/rtsp-media-factory.h:
14225 Add media factory to map urls to media pipeline objects.
14227 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14229 * gst/rtsp-server/rtsp-media.c:
14230 * gst/rtsp-server/rtsp-media.h:
14231 Add comments. Remove unused field
14233 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14235 * gst/rtsp-server/rtsp-session-pool.c:
14236 * gst/rtsp-server/rtsp-session-pool.h:
14237 Allow custom session pools to override the session id allocation algorithms Add some comments.
14239 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14241 * gst/rtsp-server/rtsp-session.h:
14244 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14246 * gst/rtsp-server/rtsp-client.c:
14247 * gst/rtsp-server/rtsp-client.h:
14248 Move the connection code in one place Add some comments
14250 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14252 * gst/rtsp-server/rtsp-server.c:
14253 * gst/rtsp-server/rtsp-server.h:
14254 Make vmethod to create and accept new clients. Add some docs.
14256 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14258 * gst/rtsp-server/rtsp-server.c:
14259 * gst/rtsp-server/rtsp-server.h:
14260 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14262 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14264 * gst/rtsp-server/rtsp-client.c:
14265 * gst/rtsp-server/rtsp-client.h:
14266 Name the parameters more appropriately.
14268 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14270 * gst/rtsp-server/rtsp-session-pool.c:
14271 Do some more cleanup of the session pool.
14273 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14275 * gst/rtsp-server/Makefile.am:
14276 * gst/rtsp-server/rtsp-client.c:
14277 Check if return value of gst_rtsp_session_get_media is not NULL
14279 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14281 * gst/rtsp-server/Makefile.am:
14282 Install rtsp-session and rtsp-session-pool headers
14284 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14289 * bindings/python/Makefile.am:
14290 * bindings/python/arg-types.py:
14291 * bindings/python/codegen/Makefile.am:
14292 * bindings/python/codegen/__init__.py:
14293 * bindings/python/codegen/argtypes.py:
14294 * bindings/python/codegen/code-coverage.py:
14295 * bindings/python/codegen/codegen.py:
14296 * bindings/python/codegen/definitions.py:
14297 * bindings/python/codegen/defsparser.py:
14298 * bindings/python/codegen/docextract.py:
14299 * bindings/python/codegen/docgen.py:
14300 * bindings/python/codegen/fileprefix.override:
14301 * bindings/python/codegen/fileprefixmodule.c:
14302 * bindings/python/codegen/h2def.py:
14303 * bindings/python/codegen/mergedefs.py:
14304 * bindings/python/codegen/mkskel.py:
14305 * bindings/python/codegen/override.py:
14306 * bindings/python/codegen/reversewrapper.py:
14307 * bindings/python/codegen/scmexpr.py:
14308 * bindings/python/rtspserver-types.defs:
14309 * bindings/python/rtspserver.defs:
14310 * bindings/python/rtspserver.override:
14311 * bindings/python/rtspservermodule.c:
14313 Add python bindings.
14315 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14317 * bindings/Makefile.am:
14319 Don't go into python dir when requirements for python bindings are missing
14321 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14323 * bindings/Makefile.am:
14324 * bindings/vala/Makefile.am:
14326 Install Vala bindings if vala is available
14328 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14330 * bindings/vala/gst-rtsp-server-0.10.deps:
14331 * bindings/vala/gst-rtsp-server-0.10.vapi:
14332 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14333 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14334 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14335 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14336 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14337 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14338 Regenerated Vala bindings
14340 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14342 * bindings/vala/gst-rtsp-server.vapi:
14343 * bindings/vala/packages/gst-rtsp-server.metadata:
14344 Fixed typo in included headers for vala bindings
14346 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14350 * pkgconfig/Makefile.am:
14351 * pkgconfig/gst-rtsp-server.pc.in:
14352 Added pkgconfig file
14354 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14356 * bindings/vala/gst-rtsp-server.vapi:
14357 * bindings/vala/packages/gst-rtsp-server.excludes:
14358 * bindings/vala/packages/gst-rtsp-server.gi:
14359 * bindings/vala/packages/gst-rtsp-server.metadata:
14360 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14362 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14364 * bindings/vala/gst-rtsp-server.vapi:
14365 * bindings/vala/packages/gst-rtsp-server.deps:
14366 * bindings/vala/packages/gst-rtsp-server.files:
14367 * bindings/vala/packages/gst-rtsp-server.gi:
14368 * bindings/vala/packages/gst-rtsp-server.metadata:
14369 * bindings/vala/packages/gst-rtsp-server.namespace:
14370 Added Vala bindings
14372 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14374 * gst/rtsp-server/rtsp-session.c:
14375 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14377 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14379 * examples/Makefile.am:
14380 * gst/rtsp-server/Makefile.am:
14381 Put GStreamer version in library name
14383 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14385 * examples/Makefile.am:
14386 * gst/rtsp-server/Makefile.am:
14387 Fix some issues to pass distcheck
14389 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14391 * gst/rtsp-server/rtsp-server.c:
14392 Added port property to GstRTSPServer class.
14394 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14399 * examples/Makefile.am:
14402 * gst/rtsp-server/Makefile.am:
14403 * gst/rtsp-server/rtsp-client.c:
14404 * gst/rtsp-server/rtsp-client.h:
14405 * gst/rtsp-server/rtsp-media.c:
14406 * gst/rtsp-server/rtsp-media.h:
14407 * gst/rtsp-server/rtsp-server.c:
14408 * gst/rtsp-server/rtsp-server.h:
14409 * gst/rtsp-server/rtsp-session-pool.c:
14410 * gst/rtsp-server/rtsp-session-pool.h:
14411 * gst/rtsp-server/rtsp-session.c:
14412 * gst/rtsp-server/rtsp-session.h:
14414 Split in library and example program
14416 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14418 * src/rtsp-client.h:
14419 Removed obsolete variable
14421 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14423 * src/rtsp-client.c:
14424 * src/rtsp-client.h:
14425 Removed pipeline variable GstRTSPClient, because it's only used in one function
14427 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14429 * src/rtsp-media.c:
14430 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14432 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14434 * src/rtsp-session.c:
14435 Initialize some more vars.
14437 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14439 * src/rtsp-session.c:
14440 Initialize variable to avoid compiler warning.
14442 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14445 Add a reasonable generic .gitignore