3 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
7 * docs/gst_plugins_cache.json:
8 * gst-rtsp-server.doap:
12 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
15 Update ChangeLogs for 1.19.3
17 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
20 meson: require matching GStreamer dep versions for unstable development releases
21 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
22 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
24 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
26 * tests/check/meson.build:
27 meson: update for meson.build_root() and .build_source() deprecation
28 -> use meson.project_build_root() or .global_build_root() instead.
29 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
31 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
34 * tests/check/meson.build:
35 meson: update for dep.get_pkgconfig_variable() deprecation
36 ... in favour of dep.get_variable('foo', ..) which in some
37 cases allows for further cleanups in future since we can
38 extract variables from pkg-config dependencies as well as
39 internal dependencies using this mechanism.
40 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
42 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
44 * gst/rtsp-server/meson.build:
45 * gst/rtsp-sink/meson.build:
46 rtsp-server: define G_LOG_DOMAIN
48 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
50 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
53 meson: bump meson requirement to >= 0.59
54 For monorepo build and ugly/bad, for advanced feature
55 option API like get_option('xyz').required(..) which
56 we use in combination with the 'gpl' option.
57 For rest of modules for consistency (people will likely
58 use newer features based on the top-level requirement).
59 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
61 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
64 meson: Streamline the way we detect when to build documentation
65 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
67 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
70 * gst/rtsp-server/meson.build:
72 meson: List libraries and their corresponding gir definition
73 Introduces a `libraries` variable that contains all libraries in a
74 list with the following format:
79 'gir': [ {full gir definition in a dict } ]
84 It therefore refactors the way we build the gir so that we can reuse the
85 same information to build them against 'gstreamer-full' in gst-build
86 when linking statically
87 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
89 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
91 * gst/rtsp-server/meson.build:
92 meson: Mark files as files()
93 Making it more robust and future proof
94 And fix issues that it creates
95 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
97 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
99 * gst/rtsp-server/rtsp-media.c:
100 rtsp-media: Unprepare suspended medias too
101 Previously suspended medias immediately reached the UNPREPARED state
102 without going through the media's unprepare() vfunc. This didn't allow
103 the media subclass to do any additional cleanup, and for example the
104 shutdown-eos property of GstRTSPMedia was ignored.
105 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
107 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
109 * gst/rtsp-server/rtsp-media.c:
110 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
111 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
113 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
115 * gst/rtsp-server/rtsp-client.c:
116 * gst/rtsp-server/rtsp-session.c:
117 * gst/rtsp-server/rtsp-session.h:
118 rtsp-client: make sure sessmedia will not get freed while used
119 handle_*_request() functions were all retrieving the session media from
120 the session by calling gst_rtsp_session_get_media () which is a transfer-none
121 call. If a session timeout happens at that time, the session media may get freed
122 making the pointer invalid..
124 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
126 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
128 * gst/rtsp-server/rtsp-media.c:
129 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
130 Previously the status was only changed for other medias.
131 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
133 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
135 * gst/rtsp-server/rtsp-session.c:
136 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
137 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
138 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
140 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
143 doc: update IRC links to OFTC
144 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
146 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
148 * docs/gst_plugins_cache.json:
151 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
153 === release 1.19.2 ===
155 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
160 * docs/gst_plugins_cache.json:
161 * gst-rtsp-server.doap:
165 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
167 * gst/rtsp-server/rtsp-media.c:
168 * gst/rtsp-server/rtsp-stream.c:
169 * gst/rtsp-server/rtsp-stream.h:
170 * gst/rtsp-sink/gstrtspclientsink.c:
171 Protection against early RTCP packets.
172 When receiving RTCP packets early the funnel is not ready yet and
173 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
174 This causes the thread that handle RTCP packets to go to pause mode.
175 Since this thread is in pause mode there will be no further callbacks to
176 handle keep-alive for incoming RTCP packets. This will make the session
177 time out if the client is not using another keep-alive mechanism.
178 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
179 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
181 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
185 Update COPYING.LIB, COPYING files
186 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
188 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
190 * docs/gst_plugins_cache.json:
194 === release 1.19.1 ===
196 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
201 * docs/gst_plugins_cache.json:
202 * gst-rtsp-server.doap:
206 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
208 * gst/rtsp-server/rtsp-stream.c:
209 rtsp-stream: use new gst_buffer_new_memdup()
210 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
212 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
214 * gst/rtsp-server/rtsp-media-factory-uri.c:
215 rtsp-media: fix leak when adding converter
216 Free the previous caps before reusing the variable for the converter caps.
217 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
219 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
221 * gst/rtsp-server/rtsp-client.c:
222 rtsp-client: fix leak adding headers
223 gst_rtsp_message_add_header() makes a copy of the header, instead
225 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
227 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
229 * gst/rtsp-server/rtsp-stream.c:
230 Use gst_element_request_pad_simple...
231 Instead of the deprecated gst_element_get_request_pad.
232 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
234 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
236 * gst/rtsp-server/rtsp-media.c:
237 rtsp-media: Ensure the bus watch is removed during unprepare
238 It's possible for the destruction of the source to be delayed.
239 Instead of relying on the dispose() to remove the bus watch, do
241 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
243 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
246 docs: minor spelling correction in README
247 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
249 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
251 * examples/test-replay-server.c:
252 test-replay-server: minor spelling corrections
253 Bumped on these while investigating the example code.
254 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
256 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
258 * tests/check/gst/stream.c:
259 tests: Don't fail tests if IPv6 not available.
260 On computers with IPv6 disabled it shouldn't result in a test failure.
261 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
263 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
265 * gst/rtsp-server/rtsp-media.c:
266 rtsp-media: Add one more case to seek avoidance
267 This is an extension to the previous commit. There can also be cases where the
268 start position is not specified, in those cases we should also avoid doing
269 seeking unless it's forced.
270 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
272 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
274 * gst/rtsp-server/rtsp-media.c:
275 rtsp-media: Improve skipping trickmode seek.
276 We can also skip the seek if the end range is already
278 Avoids initial seek on play start if playing full stream.
279 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
281 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
283 * gst/rtsp-sink/gstrtspclientsink.c:
284 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
285 It's sufficient to run them during the FIRST stage instead of in both.
286 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
288 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
290 * tests/check/gst/rtspclientsink.c:
291 tests: rtspclientsink: fix some leaks
292 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
294 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
296 * gst/rtsp-sink/gstrtspclientsink.c:
297 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
298 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
300 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
302 * tests/check/gst/rtspclientsink.c:
303 rtspclientsink: add unit test for potential shutdown deadlock
304 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
306 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
308 * gst/rtsp-sink/gstrtspclientsink.c:
309 rtspclientsink: fix deadlock on shutdown before preroll
310 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
311 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
313 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
315 * gst/rtsp-server/rtsp-stream.c:
316 rtsp-stream: avoid deadlock in send_func
317 Currently the send_func() runs in a thread of its own which is started
318 the first time we enter handle_new_sample(). It runs in an outer loop
319 until priv->continue_sending is FALSE, which happens when a TEARDOWN
320 request is received. We use a local variable, cont, which is initialized
321 to TRUE, meaning that we will always enter the outer loop, and at the
322 end of the outer loop we assign it the value of priv->continue_sending.
323 Within the outer loop there is an inner loop, where we wait to be
324 signaled when there is more data to send. The inner loop is exited when
325 priv->send_cookie has changed value, which it does when more data is
326 available or when a TEARDOWN has been received.
327 But if we get a TEARDOWN before send_func() is entered we will get stuck
328 in the inner loop because no one will increase priv->session_cookie
330 By not entering the outer loop in send_func() if priv->continue_sending
331 is FALSE we make sure that we do not get stuck in send_func()'s inner
332 loop should we receive a TEARDOWN before the send thread has started.
333 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
334 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
336 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
338 * gst/rtsp-server/rtsp-client.c:
339 rtsp-client: cleanup transports during TEARDOWN
340 When tunneling RTP over RTSP the stream transports are stored in a hash
341 table in the GstRTSPClientPrivate struct. They are used for, among other
342 things, mapping channel id to stream transports when receiving data from
343 the client. The stream tranports are created and added to the hash table
344 in handle_setup_request(), but unfortuately they are not removed in
345 handle_teardown_request(). This means that if the client sends data on
346 the RTSP connection after it has sent the TEARDOWN, which is often the
347 case when audio backchannel is enabled, handle_data() will still be able
348 to map the channel to a session transport and pass the data along to it.
349 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
350 because the stream is no longer joined to a bin.
351 We avoid this by removing the stream transports from the hash table when
352 we handle the TEARDOWN request.
353 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
355 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
357 * docs/gst_plugins_cache.json:
358 * gst/rtsp-sink/gstrtspclientsink.c:
359 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
360 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
362 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
364 * tests/check/gst/client.c:
365 Add test cases for mountpoint of '/'
366 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
368 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
370 * gst/rtsp-server/rtsp-client.c:
371 * gst/rtsp-server/rtsp-mount-points.c:
372 * gst/rtsp-server/rtsp-session-media.c:
373 Make a mount point of "/" work correctly.
374 As far as I can tell, this is neither explicitly allowed nor
375 forbidden by RFC 7826.
376 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
377 use in the wild (presumably with non-GStreamer servers).
378 GStreamer's prior behavior was confusing, in that
379 gst_rtsp_mount_points_add_factory() would appear to accept a mount
380 path of "" or "/", but later connection attempts would fail with a
381 "media not found" error.
382 This commit makes a mount path of "/" work for either form of URL,
383 while an empty mount path ("") is rejected and logs a warning.
384 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
386 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
388 * docs/gst_plugins_cache.json:
389 * gst/rtsp-sink/gstrtspclientsink.c:
390 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
391 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
393 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
395 * gst/rtsp-server/rtsp-media.c:
396 rtsp-media: Only count senders when counting blocked streams
397 Only sender streams sends the GstRTSPStreamBlocking message, so only
398 these should be counted before setting media status to prepared.
399 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
401 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
403 * gst/rtsp-sink/gstrtspclientsink.c:
404 rtspclientsink add proper support for uri queries
405 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
407 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
409 * gst/rtsp-server/rtsp-client.c:
410 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
411 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
412 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
414 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
416 * gst/rtsp-server/rtsp-stream.c:
417 rtsp-stream: collect a clock_rate when blocking
418 This lets us provide a clock_rate in a fashion similar to the
419 other code paths in get_rtpinfo()
420 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
422 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
424 * gst/rtsp-server/rtsp-media.c:
425 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
426 Otherwise this will cause memory corruption as the property expects a 64
428 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
430 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
432 * gst/rtsp-server/rtsp-media.c:
433 * gst/rtsp-server/rtsp-stream.c:
434 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
435 To prevent cases with prerolling when the inactive stream prerolls first
436 and the server proceeds without waiting for the active stream, we will
437 ignore GstRTSPStreamBlocking messages from incomplete streams. When
438 there are no complete streams (during DESCRIBE), we will listen to all
440 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
442 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
444 * tests/check/gst/media.c:
445 * tests/check/meson.build:
446 * tests/files/test.avi:
447 media test: Add test for seeking one active stream with a demuxer
448 Add another seek_one_active_stream test but with a demuxer. The demuxer
449 will flush both streams in opposed to the existing test which only
450 flushes the active stream. This will help exposing problems with the
451 prerolling process after a flushing seek.
452 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
454 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
456 * gst/rtsp-server/meson.build:
458 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
459 * pkgconfig/gstreamer-rtsp-server.pc.in:
460 * pkgconfig/meson.build:
461 Meson: Use pkg-config generator
462 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
464 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
467 meson: update glib minimum version to 2.56
468 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
470 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
472 * examples/test-launch.c:
473 * gst/rtsp-server/rtsp-media-factory.c:
474 * gst/rtsp-server/rtsp-media-factory.h:
475 * gst/rtsp-server/rtsp-media.c:
476 * gst/rtsp-server/rtsp-server-internal.h:
477 * gst/rtsp-server/rtsp-stream.c:
478 * tests/check/gst/client.c:
479 rtsp-media-factory: expose API to disable RTCP
480 This is supported by the RFC, and can be useful on systems where
481 allocating two consecutive ports is problematic, and RTCP is not
483 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
485 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
487 * hooks/pre-commit.hook:
489 git: use our standard pre commit hook
490 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
492 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
494 * gst/rtsp-server/rtsp-stream.c:
495 rtsp-stream: make use of blocked_running_time in query_position
496 When blocking, the sink element will not have received a buffer
497 yet and the position query will fail. Instead, we make use of
498 the running time of the buffer we blocked on.
499 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
501 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
503 * gst/rtsp-server/rtsp-stream.c:
504 rtsp-stream: collect rtp info when blocking
505 We don't unblock the stream anymore before replying to the
506 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
507 so the sinks don't have a last-sample after potentially flush
508 seeking. seek_trickmode waits for preroll however, which means
509 the stream will block and wait for a first buffer. Subsequent
510 calls to get_rtpinfo() can thus make use of the information.
511 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
512 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
514 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
516 * examples/meson.build:
517 * examples/test-replay-server.c:
518 * examples/test-replay-server.h:
519 examples: Add an example for loop playback
520 This demo example shows a way of file loop playback of a given source.
521 Note that client seek request is not properly implemented yet.
522 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
524 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
526 * gst/rtsp-server/rtsp-media.c:
527 rtsp-media: Plug memory leak
528 The get-storage signal of rtpbin increases the ref count of the storage.
529 So we have to unref it after usage.
530 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
532 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
534 * gst/rtsp-server/rtsp-media.c:
535 rtsp-media: Get rates only on sender streams
536 When play a media with both sender and receiver stream, like ONVIF
537 back channel audio in, gst_rtsp_media_get_rates call
538 gst_rtsp_stream_get_rates for each stream to set the rates. But
539 gst_rtsp_stream_get_rates return false for the receiver steam, which
540 lead a g_assert crash.
541 Instead to get rates on all streams, now just get rates on sender
543 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
545 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
547 * gst/rtsp-server/rtsp-media.c:
548 * gst/rtsp-server/rtsp-server-internal.h:
549 * gst/rtsp-server/rtsp-stream.c:
550 rtsp-media: set a 0 storage size for TCP receivers
551 ulpfec correction is obviously useless when receiving a stream
552 over TCP, and in TCP modes the rtp storage receives non
553 timestamped buffers, causing it to queue buffers indefinitely,
554 until the queue grows so large that sanity checks kick in and
555 warnings start to get emitted.
556 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
558 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
560 * gst/rtsp-server/rtsp-stream.c:
561 rtsp-stream: preroll on gap events
562 This allows negotiating a SDP with all streams present, but only
563 start sending packets at some later point in time
564 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
566 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
568 * gst/rtsp-server/rtsp-media.c:
569 rtsp-media: do not unblock on unsuspend
570 rtsp_media_unsuspend() is called from handle_play_request()
571 before sending the play response. Unblocking the streams here
572 was causing data to be sent out before the client was ready
573 to handle it, with obvious side effects such as initial packets
574 getting discarded, causing decoding errors.
575 Instead we can simply let the media streams be unblocked when
576 the state of the media is set to PLAYING, which occurs after
577 sending the play response.
578 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
580 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
583 ci: include template from gst-ci master branch again
585 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
587 * docs/gst_plugins_cache.json:
591 === release 1.18.0 ===
593 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
599 * docs/gst_plugins_cache.json:
600 * gst-rtsp-server.doap:
604 === release 1.17.90 ===
606 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
611 * docs/gst_plugins_cache.json:
612 * gst-rtsp-server.doap:
616 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
618 * gst/rtsp-server/rtsp-thread-pool.c:
619 rtsp-thread-pool.c: fix clang 10 warning
620 clang 10 is complaining about incompatible types due to the
623 ../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]
625 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
627 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
629 * gst/rtsp-server/rtsp-thread-pool.c:
630 rtsp-thread-pool.c: fix clang 10 warning
631 clang 10 is complaining about incompatible types due to the
634 ../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]
636 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
638 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
640 * gst/rtsp-server/rtsp-sdp.c:
641 rtsp-sdp: Fix resource leak in mikey messsage
642 Fixed a resource leak for mikey message while adding crypto session
644 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
646 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
649 * scripts/extract-release-date-from-doap-file.py:
650 meson: set release date from .doap file for releases
651 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
653 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
655 * gst/rtsp-server/rtsp-stream.c:
656 rtsp-stream: explicitly set caps on udpsrc elements
657 This causes them to send caps events before data flow, which is
658 usually a pretty correct thing to do!
659 Not doing so manifested in a bug where ssrcdemux wouldn't forward
660 the caps it had received with an extra ssrc field, as it hadn't
661 received any caps event.
663 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
665 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
667 * docs/gst_plugins_cache.json:
671 === release 1.17.2 ===
673 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
678 * docs/gst_plugins_cache.json:
679 * gst-rtsp-server.doap:
683 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
685 * docs/gst_plugins_cache.json:
686 doc: Stop documenting properties from parents
688 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
690 * docs/gst_plugins_cache.json:
691 docs: Fix version in the plugins cache
692 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
694 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
696 * gst/rtsp-sink/gstrtspclientsink.c:
697 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
698 It's deprecated, unneeded and doesn't do anything anymore.
699 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
701 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
706 === release 1.17.1 ===
708 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
713 * docs/gst_plugins_cache.json:
714 * gst-rtsp-server.doap:
718 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
720 * gst/rtsp-server/rtsp-media.c:
721 rtsp-media: Add/configure transports when completing the pipeline
722 Otherwise the transports are not set up yet during the PLAY request
723 handling when unsuspending (and thus unblocking) the media.
724 In case of live pipelines this then causes the first few packets to go
725 to the sinks before they know what to do with them, and they simply
726 discard them which is rather suboptimal in case of keyframes.
727 For non-live pipelines this is not a problem because the sink will still
728 be PAUSED and as such not send out the data yet but wait until it goes
729 to PLAYING, which is late enough.
730 Adding the transports multiple times is not a problem: if the transport
731 is already added it won't be added another time and TRUE will be
733 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
735 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
736 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
738 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
740 * gst/rtsp-server/rtsp-media.c:
741 rtsp-media: Fix misleading comment
742 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
744 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
746 * gst/rtsp-server/rtsp-media.c:
747 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
748 The pad probes are not needed anymore at this point and later when
749 reaching buffering 100% only the state is changed, no unblocking
751 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
753 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
755 * gst/rtsp-server/rtsp-media.c:
756 rtsp-media: Remove duplicated media_unblock() function
757 It does literally the same as media_streams_set_blocked(FALSE).
758 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
760 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
762 * examples/test-onvif-server.c:
763 test-onvif-server: cast ntp-offset property value to 64 bit
764 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
766 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
768 * docs/gst_plugins_cache.json:
769 docs: Update plugins cache
771 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
773 * examples/test-onvif-server.c:
774 * examples/test-onvif-server.h:
775 * gst/rtsp-server/rtsp-onvif-media-factory.h:
776 onvif-media-factory: define autoptr cleanup function
777 And have the factory in the onvif-server example inherit from
778 GstRTSPOnvifMediaFactory.
779 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
781 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
783 * docs/gst_plugins_cache.json:
784 docs: Update plugins cache
786 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
788 * tests/check/gst/rtspserver.c:
789 tests: enforce I420 format
790 Test was not enforcing a video format on videotestsrc. I420 was picked as it
791 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
792 true (gst-plugins-base!689).
793 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
795 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
797 * gst/rtsp-sink/gstrtspclientsink.c:
798 plugins: uddate gst_type_mark_as_plugin_api() calls
800 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
803 doc: Require hotdoc >= 0.11.0
805 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
807 * docs/gst_plugins_cache.json:
808 docs: Update gst_plugins_cache.json
810 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
812 * gst/rtsp-sink/gstrtspclientsink.c:
813 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
815 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
817 * gst/rtsp-server/meson.build:
818 meson: gir: remove bogus sources_top_dir kwarg
819 Doesn't actually exist. Was fixed differently in Meson
820 so that the user doesn't have to specify it.
821 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
823 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
825 * tests/check/meson.build:
826 tests: put registry into tests/check not the gst/ subdir
827 Underscorify the test name before setting GST_REGISTRY,
828 so the registry actually ends up in the current build dir
830 For consistency with the other modules, but should also
831 avoid problems on windows.
832 Also fix indentation of environment block.
833 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
835 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
837 * tests/check/meson.build:
838 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
839 If core is built as a subproject (e.g. as in gst-build), make sure to use
840 the gst-plugin-scanner from the built subproject. Without this, gstreamer
841 might accidentally use the gst-plugin-scanner from the install prefix if
842 that exists, which in turn might drag in gst library versions we didn't
843 mean to drag in. Those gst library versions might then be older than
844 what our current build needs, and might cause our newly-built plugins
845 to get blacklisted in the test registry because they rely on a symbol
846 that the wrongly-pulled in gst lib doesn't have.
847 This should fix running of unit tests in gst-build when invoking
848 meson test or ninja test from outside the devenv for the case where
849 there is an older or different-version gst-plugin-scanner installed
850 in the install prefix.
851 In case no gst-plugin-scanner is installed in the install prefix, this
852 will fix "GStreamer-WARNING: External plugin loader failed. This most
853 likely means that the plugin loader helper binary was not found or
854 could not be run. You might need to set the GST_PLUGIN_SCANNER
855 environment variable if your setup is unusual." warnings when running
857 In the case where we find GStreamer core via pkg-config we use
858 a newly-added pkg-config var "pluginscannerdir" to get the right
859 directory. This has the benefit of working transparently for both
860 installed and uninstalled pkg-config files/setups.
861 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
863 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
865 * tests/check/meson.build:
866 tests: gst-plugins-base and -bad plugins are required for the unit tests
867 Make hard requirement until we have more fine-grained control
868 in the unit tests. Of course the presence of the .pc file doesn't
869 imply that the plugins we need are actually there, but it's at
870 least a step in the right direction.
871 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
873 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
875 * tests/check/meson.build:
876 tests: pick up rtsp-server plugins from build directory only
877 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
879 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
881 * gst/rtsp-server/rtsp-media.c:
882 rtsp-media: wait for all GstRTSPStreamBlocking messages
883 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
884 each active stream when checking if all streams are blocked.
885 Without this change there will be a race condition when using two or
886 more streams and rtsp-media receives a GstRTSPStreamBlocking message
887 from one of the streams. This is because rtsp-media then checks if all
888 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
889 stream. This function call returns TRUE if the stream has sent a
890 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
891 receive this message. This would then result in that rtsp-media
892 erroneously thinks it is blocking all streams which could result in
893 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
894 preroll, this could result in that rtsp-media thinks that the pipeline
895 is prerolled even though that might not be the case.
896 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
898 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
900 * gst/rtsp-server/rtsp-media.c:
901 rtsp-media: update expected_async_done during suspend
902 Set expected_async_done to FALSE in default_suspend() if a state change
903 occurs and the return value from set_target_state() is something other
904 than GST_STATE_CHANGE_ASYNC.
905 Without this change there is a risk that expected_async_done will be
906 TRUE even though no asynchronous state change is taking place. This
907 could happen if the pipeline is set to PAUSED using
908 media_set_pipeline_state_locked(), an asynchronous state change starts
909 and then the media is suspended (which could result in a state change,
910 aborting the asynchronous state change). If the media is suspended
911 before the asynchronous state change ends then expected_async_done will
912 be TRUE but no asynchronous state change is taking place.
913 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
915 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
917 * gst/rtsp-server/rtsp-client.c:
918 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
919 There was a race condition where client was being finalized and
920 concurrently in some other thread the rtsp ctrl timout was relying on
921 client data that was being freed.
922 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
923 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
925 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
927 * gst/rtsp-server/rtsp-media-factory.c:
928 * gst/rtsp-server/rtsp-media-factory.h:
929 * gst/rtsp-server/rtsp-media.c:
930 * gst/rtsp-server/rtsp-media.h:
931 media-factory: complete DSCP QoS setting support
932 add dscp_qos setting support at factory and media level to setup IP DSCP
933 field of bounded UDP sinks.
934 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
935 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
937 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
939 * gst/rtsp-server/rtsp-client.c:
940 rtsp-client: Fix some race conditions around timeout source removal
941 We always need to take the lock while accessing it as otherwise another
942 thread might've removed it in the meantime. Also when destroying and
943 creating a new one, ensure that the mutex is not shortly unlocked in
944 between as during that time another one might potentially be created
946 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
948 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
950 * gst/rtsp-server/rtsp-media.c:
951 * gst/rtsp-server/rtsp-stream.c:
952 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
953 And the same for gst_rtsp_stream_get_rates().
954 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
956 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
958 * examples/test-onvif-server.c:
959 examples: test-onvif-server: fix compiler warnings on raspbian
960 Fix printf format for 64-bit variables.
961 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
963 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
965 * gst/rtsp-server/rtsp-stream-transport.c:
966 * gst/rtsp-server/rtsp-stream-transport.h:
967 * gst/rtsp-server/rtsp-stream.c:
968 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
969 The old API is preserved now and new API was added that provides the
970 additional parameter to the callback.
971 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
972 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
974 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
976 * gst/rtsp-server/rtsp-client.c:
977 rtsp-client: Store the timeout source by pointer instead of id
978 That way we don't have to retrieve it again from the main context when
979 destroying it but can directly do so.
980 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
982 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
984 * gst/rtsp-server/rtsp-client.c:
985 rtsp-client: Clean up watch/watch context and related state consistently
986 And assert that it was cleaned up properly before the client is
987 finalized. If something is still around when the client is shut down
988 then something went very wrong before.
989 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
991 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
993 * gst/rtsp-server/rtsp-client.c:
994 * tests/check/gst/rtspserver.c:
995 rtsp-client: Combine the pre-session and post-session timeout
996 They previously used the same state but different mechanisms and
997 functions, which was difficult to follow, error prone and simply
999 Also adjust the test for the post-session timeout a bit to be less racy
1000 now that the timing has slightly changed.
1001 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1003 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1005 * gst/rtsp-server/rtsp-client.c:
1006 rtsp-client: Don't ever close the client connection directly when a session is torn down
1007 There might be other sessions that are running over the same RTSP
1008 connection and we should not simply close the client directly if one of
1010 By default the connection will be closed once the client closes it or
1011 the OS does. This behaviour can be adjusted with the
1012 post-session-timeout property, which allows to close it automatically
1013 from the server side after all sessions are gone and the given timeout
1015 This reverts the previous commit.
1016 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1018 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1020 * gst/rtsp-server/rtsp-client.c:
1021 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1022 Instead of closing it never at all. Previously there was only code that
1023 closed the client asynchronously if sending the response happened
1024 asynchrously at a later time.
1025 Thanks to Christian M for debugging this issue.
1026 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1027 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1029 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1031 * gst/rtsp-server/rtsp-stream.c:
1032 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1033 Otherwise no sink is found for multicast sreams and the less accurate
1034 fallback is used to determine the current sequence number and timestamp.
1036 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1038 * gst/rtsp-server/rtsp-auth.c:
1039 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1040 When using the basic authentication scheme, we wouldn't validate that
1041 the authorization field of the credentials is not NULL and pass it on
1042 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1043 dereference the NULL pointer and crash.
1044 A specially crafted (read: invalid) RTSP header can cause this to
1046 As a solution, check for the authorization to be not NULL before
1047 continuing processing it and if it is simply fail authentication.
1048 This fixes CVE-2020-6095 and TALOS-2020-1018.
1049 Discovered by Peter Wang of Cisco ASIG.
1051 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1053 * gst/rtsp-server/rtsp-client.c:
1054 rtsp-client: Use watch_context before unref
1055 Move the usage of priv->watch_context to beginning of function
1056 gst_rtsp_client_finalize. Instead of use it after
1057 g_main_context_unref (priv->watch_context).
1059 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1061 * gst/rtsp-server/rtsp-stream.c:
1062 rtsp-stream: fix deadlock on transport removal
1063 We cannot take the RTSPStream lock while holding a transport backlog
1064 lock, as remove_transport may be called externally, which will
1065 take first the RTSPStream lock then the transport backlog lock.
1067 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1069 * gst/rtsp-server/rtsp-server-internal.h:
1070 * gst/rtsp-server/rtsp-stream-transport.c:
1071 * gst/rtsp-server/rtsp-stream.c:
1072 rtsp-stream: clear backlog when removing transport
1073 This ensures we don't end up calling any of transports' callbacks
1074 with a potentially unreffed user_data (in practice, a client that
1075 may have been removed)
1077 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1079 * gst/rtsp-server/rtsp-stream.c:
1080 rtsp-stream: marshal calls to send_tcp_message to a single thread
1081 In order to address the race condition pointed out at
1082 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1083 we get rid of the send thread pool, and instead spawn and manage
1084 a single thread to pull samples from app sinks and add them to
1085 the transport's backlogs.
1086 Additionally, we now also always go through the backlogs in order
1087 to simplify the logic.
1089 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1091 * gst/rtsp-server/rtsp-server-internal.h:
1092 * gst/rtsp-server/rtsp-stream-transport.c:
1093 * gst/rtsp-server/rtsp-stream.c:
1094 rtsp-stream: properly protect TCP backlog access
1096 We cannot hold stream->lock while pushing data, but need
1097 to consistently check the state of the backlog both from
1098 the send_tcp_message function and the on_message_sent function,
1099 which may or may not be called from the same thread.
1100 This commit introduces internal API to allow for potentially
1101 recursive locking of transport streams, addressing a race
1102 condition where the RTSP stream could push items out of order
1103 when popping them from the backlog.
1105 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1107 * gst/rtsp-server/rtsp-media.c:
1108 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1109 It's taken ownership of by the media, and returned with `transfer none`
1110 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1111 first then any bindings will wrongly take ownership of the pipeline once
1112 it arrives in bindings code.
1114 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1116 * examples/test-onvif-client.c:
1117 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1119 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1121 * gst/rtsp-server/rtsp-media.c:
1122 rtsp-media: fix default latency
1124 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1126 * gst/rtsp-server/rtsp-client.c:
1127 rtsp-client: make closing more thread safe
1128 + Take the watch lock prior to using priv->watch
1129 + Flush both the watch and connection before closing / unreffing
1130 gst_rtsp_connection_close() is not threadsafe on its own, this is
1131 a workaround at the client level, where we control both the watch
1134 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1136 * gst/rtsp-server/rtsp-latency-bin.c:
1137 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1140 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1141 `your_type_get_instance_private()` function instead
1144 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1146 * gst/rtsp-server/rtsp-client.c:
1147 * tests/check/gst/rtspserver.c:
1148 rtsp-client: add property post-session-timeout
1149 This is a TCP connection timeout for client connections, in seconds.
1150 If a positive value is set for this property, the client connection
1151 will be kept alive for this amount of seconds after the last session
1152 timeout. For negative values of this property the connection timeout
1153 handling is delegated to the system (just as it was before).
1156 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1158 * gst/rtsp-server/rtsp-stream.c:
1159 rtsp-stream: check for NULL transports prior to ref'ing
1161 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1163 * gst/rtsp-server/rtsp-server-internal.h:
1164 * gst/rtsp-server/rtsp-stream-transport.c:
1165 * gst/rtsp-server/rtsp-stream.c:
1166 rtsp-stream: fix checking of TCP backpressure
1167 The internal index of our appsinks, while it can be used to
1168 determine whether a message is RTP or RTCP, is not necessarily
1169 the same as the interleaved channel. Let the stream-transport
1170 determine the channel to check backpressure for, the same way
1171 it determines the channel according to whether it is sending
1174 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1176 * gst/rtsp-server/rtsp-session.c:
1177 rtsp-session: Butcher the file to please gst-indent in the CI
1178 This should be reverted once the CI has an updated gst-indent.
1180 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1182 * gst/rtsp-server/rtsp-session.c:
1183 * gst/rtsp-server/rtsp-session.h:
1184 * gst/rtsp-sink/gstrtspclientsink.c:
1185 * gst/rtsp-sink/gstrtspclientsink.h:
1186 rtsp-session & client: Remove deprecated GTimeVal
1187 GTimeVal won't work past 2038
1189 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1191 * gst/rtsp-server/rtsp-auth.c:
1192 rtsp-auth: fix default token leak
1194 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1196 * gst/rtsp-sink/gstrtspclientsink.c:
1197 gstrtspclientsink: unref transports when closing bin
1200 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1202 * gst/rtsp-server/rtsp-media.c:
1203 rtsp-media: Force seek when flush flag is set
1204 The commit "rtsp-client: define all seek accuracy flags from
1205 setup_play_mode" changed the behaviour of when doing a seek.
1206 Before that commit, having the flush flag set would result in a seek
1208 Even if no seek was needed. One reason to force seek is to flush old buffers
1209 created in Describe requests.
1210 Thus adding force seek also for flush flag will result in play request
1213 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1215 * gst/rtsp-server/rtsp-client.c:
1216 rtsp-client: Revitalize dead code
1217 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1220 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1222 * gst/rtsp-server/rtsp-sdp.c:
1223 rtsp-sdp: Don't try to use non-initialized values
1224 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1225 returns TRUE. Also avoid the whole clock signalling block if we're not
1226 dealing with senders.
1231 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1233 * gst/rtsp-server/rtsp-stream-transport.c:
1234 * gst/rtsp-server/rtsp-stream.c:
1235 * tests/check/gst/stream.c:
1236 rtsp-stream: Removing invalid transports returns false
1237 When removing transports an assertion was that the transports passed in
1238 for removal are present in the list, however that can't be assumed.
1239 As an example if a transport was removed from a thread running
1240 send_tcp_message, the main thread can try to remove the same transport
1241 again if it gets a handle_pause_request. This will not effect the
1242 transport list but it will effect n_tcp_transports as it will be
1243 decrement and then have the wrong value.
1245 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1247 * tests/check/gst/client.c:
1248 client test: add scale and speed negative tests
1249 Negative tests for scale and speed should be done as well, verify that
1250 the response code is "400 Bad request" when a bad request is done.
1252 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1254 * gst/rtsp-server/rtsp-auth.c:
1255 * gst/rtsp-server/rtsp-client.c:
1256 * gst/rtsp-server/rtsp-media-factory.c:
1257 * gst/rtsp-server/rtsp-media.c:
1258 * gst/rtsp-server/rtsp-server.c:
1259 * gst/rtsp-server/rtsp-session-pool.c:
1260 * gst/rtsp-server/rtsp-stream.c:
1261 * gst/rtsp-sink/gstrtspclientsink.c:
1262 Don't pass default GLib marshallers for signals
1263 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1264 actually internally optimize the signal (if the marshaller is available
1265 in GLib itself) by also setting the valist marshaller. This makes the
1266 signal emission a bit more performant than the regular marshalling,
1267 which still needs to box into `GValue` and call libffi in case of a
1269 Note that for custom marshallers, one would use
1270 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1272 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1274 * gst/rtsp-server/rtsp-mount-points.c:
1275 GstRTSPMountPoints: Remove any existing factory before adding a new one
1276 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1277 previous mount point will be freed" which was true when it was
1278 implemented using a GHashTable. But in 2012 it got rewrote using a
1279 GSequence and since then it could have 2 factories for the same path.
1280 Which one gets used is random, depending on the sorting order of 2
1283 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1285 * gst/rtsp-server/rtsp-client.c:
1286 * gst/rtsp-server/rtsp-server-internal.h:
1287 * gst/rtsp-server/rtsp-stream-transport.c:
1288 * gst/rtsp-server/rtsp-stream-transport.h:
1289 * gst/rtsp-server/rtsp-stream.c:
1290 stream: refactor TCP backpressure handling
1291 The previous implementation stopped sending TCP messages to
1292 all clients when a single one stopped consuming them, which
1293 obviously created problems for shared media.
1294 Instead, we now manage a backlog in stream-transport, and slow
1295 clients are removed once this backlog exceeds a maximum duration,
1296 currently hardcoded.
1299 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1302 meson: build gir even when cross-compiling if introspection was enabled explicitly
1303 This can be made to work in certain circumstances when
1304 cross-compiling, so default to not building g-i stuff
1305 when cross-compiling, but allow it if introspection was
1306 enabled explicitly via -Dintrospection=enabled.
1307 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1309 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1311 * gst/rtsp-server/rtsp-session.c:
1312 rtsp-session: clean up comment extra-timeout
1314 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1316 * gst/rtsp-server/rtsp-client.c:
1317 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1318 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1319 from the RTSP context.
1322 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1324 * gst/rtsp-server/rtsp-client.c:
1325 * gst/rtsp-server/rtsp-media.c:
1326 * gst/rtsp-server/rtsp-media.h:
1327 rtsp-client: Lock shared media
1328 For shared media we got race conditions. Concurrently rtsp clients might
1329 suspend or unsuspend the shared media and thus change the state without
1330 the clients expecting that.
1331 By introducing a lock that can be taken by callers such as rtsp_client
1332 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1333 to handle the media sequentially thus allowing one client to finish its
1334 rtsp call before another client calls on the same media.
1335 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1338 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1340 * gst/rtsp-server/rtsp-session.c:
1341 rtsp-session: add property extra-timeout
1342 Extra time to add to the timeout, in seconds. This only
1343 affects the time until a session is considered timed out
1344 and is not signalled in the RTSP request responses.
1345 Only the value of the timeout property is signalled in the
1348 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1350 * gst/rtsp-server/rtsp-stream.c:
1351 rtsp-stream : fix race condition in send_tcp_message
1352 If one thread is inside the send_tcp_message function and are done
1353 sending rtp or rtcp messages so the n_outstanding variable is zero
1354 however have not exit the loop sending the messages. While sending its
1355 messages, transports have been added or removed to the transport list,
1356 so the cache should be updated. If now an additional thread comes to
1357 the function send_tcp_message and trying to send rtp messages it will
1358 first destroy the rtp cache that is still being iterated trough by the
1362 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1371 * examples/.gitignore:
1372 * examples/Makefile.am:
1374 * gst/rtsp-server/.gitignore:
1375 * gst/rtsp-server/Makefile.am:
1376 * gst/rtsp-sink/Makefile.am:
1377 * pkgconfig/.gitignore:
1378 * pkgconfig/Makefile.am:
1380 * tests/Makefile.am:
1381 * tests/check/Makefile.am:
1382 Remove autotools build
1384 Maybe we can now use the meson pkgconfig module
1385 for .pc files? (Does it support uninstalled now?)
1387 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1389 * tests/check/gst/client.c:
1390 client: fix test mem leak in attach_rate_tweaking_probe
1392 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1394 * tests/check/gst/media.c:
1395 media: remove memleak in test test_media_seek
1397 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1399 * tests/check/gst/rtspserver.c:
1400 rtspserver: Remove memleak in test test_double_play
1402 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1404 * gst/rtsp-server/rtsp-media.c:
1405 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1407 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1409 * gst/rtsp-server/rtsp-media.c:
1410 * tests/check/gst/rtspserver.c:
1411 rtsp-media: Unblock all streams
1412 When unsuspending and going to PLAYING, unblock all streams instead of
1413 only those that are linked (the linked streams are the ones for which
1414 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1415 pushing buffers on unlinked streams.
1416 This change is because playback using single-threaded demuxers like
1417 matroska-demux could be blocked if SETUP was not called for all media.
1418 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1419 gstflvdemux, qtdemux, and matroska-demux) will handle
1420 GST_FLOW_NOT_LINKED automatically.
1423 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1425 * gst/rtsp-server/rtsp-media.c:
1426 * tests/check/gst/rtspserver.c:
1427 rtsp-media: Wait on async when needed.
1428 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1429 In the unit test the pause from adjust_play_mode will cause a preroll
1430 and after that async-done will be produced.
1431 Without this patch there are no one consuming this async-done and when
1432 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1433 wait for async-done. But then it wrongly find the async-done prodused by
1434 adjus_play_mode and continue executing without waiting for the preroll
1437 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1439 * gst/rtsp-server/rtsp-client.c:
1440 rtsp-client: RTP Info when completed_sender
1441 Change condition that should be fulfilled regarding RTPInfo.
1442 Replace !gst_rtsp_media_is_receive_only with
1443 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1444 for a sender pipeline that is complete. Only then a RTPInfo should
1446 gst_rtsp_media_is_receive_only gives different answears depending on
1448 If Describe is called wth URL+options for backchannel SDP will give only
1449 audio and only backchannel a=sendonly
1450 If Describe is called on URL+options that gives both audio and video
1451 direction from server to client, pipelines are created. Thus
1452 receive_only will return false, even though Setup only would setup
1454 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1455 streams are complete.
1457 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1459 * gst/rtsp-server/rtsp-client.c:
1460 * tests/check/gst/client.c:
1461 rtsp-client: RTP Info exists conditionally in PLAY
1462 If RTP Info is missing and it is not a receiver only, eg. audio
1463 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1464 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1465 Since 1.14 there is audio backchannel support. Thus RTP-info is
1466 conditional now. When audio backchannel only mode, there is no RTP-info.
1469 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1471 * examples/test-onvif-client.c:
1472 test-onvif-client: remove unused query
1474 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1476 * gst/rtsp-server/rtsp-client.c:
1477 rtsp-client: RTP Info must exist in PLAY response
1478 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1481 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1483 * examples/test-onvif-client.c:
1484 test-onvif-client: perform accurate seeks
1485 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1486 Also, modify how we compute the position: position queries in
1487 PAUSED mode fail to account for the newly-prerolled frame, leading
1488 to frame skips when performing seeks in that state. Instead,
1489 compute the current position from the last sample.
1491 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1493 * gst/rtsp-server/rtsp-client.c:
1494 * gst/rtsp-server/rtsp-media.c:
1495 * gst/rtsp-server/rtsp-media.h:
1496 * tests/check/gst/rtspserver.c:
1497 Use complete streams for scale and speed.
1498 Without this patch it's always stream0 that is used to get segment event
1499 that is used to set scale and speed. This even if client not doing SETUP
1500 for stream0. At least in suspend mode reset this not working since then
1501 it's just random if send_rtp_sink have got any segment event. There are
1502 no check if send_rtp_sink for stream0 got any data before media is
1503 prerolled after PLAY request.
1505 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1507 * examples/test-onvif-server.c:
1508 * examples/test-onvif-server.h:
1509 examples/onvif-server: fix werror build with clang
1510 ../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]
1511 self->incoming_segment->format, self->incoming_segment->flags,
1512 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1513 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1514 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
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>:77:1: note: expanded from here
1522 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1523 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1525 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1526 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1528 <scratch space>:9:1: note: expanded from here
1531 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1532 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1533 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1535 <scratch space>:12:1: note: expanded from here
1539 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1542 meson: Don't generate doc cache when no plugins are enabled
1543 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1545 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1547 * examples/test-onvif-client.c:
1548 test-onvif-client: stdin is not defined in MSVC
1550 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1552 * gst/rtsp-server/rtsp-media.c:
1553 rtsp-media: add missing Since tag
1555 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1557 * examples/test-onvif-client.c:
1558 test-onvif-client: STDIN_FILENO is not portable
1559 If not defined, define it to _fileno(stdin) on Windows, 0
1562 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1564 * examples/test-onvif-server.c:
1565 test-onvif-server: downgrade logging
1567 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1569 * examples/meson.build:
1570 * examples/test-onvif-client.c:
1571 * examples/test-onvif-server.c:
1572 examples: add ONVIF client / server example
1574 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1576 * gst/rtsp-server/rtsp-client.c:
1577 * gst/rtsp-server/rtsp-media.c:
1578 rtsp-client: define all seek accuracy flags from setup_play_mode
1579 We then pass those to adjust_play_mode, which needs to operate
1580 on the "final" seek flags, as previously the code in rtsp-media
1581 was assuming that accuracy seek flags (accurate / key_unit) should
1582 not be set if the flags passed to the seek method were already set.
1584 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1586 * gst/rtsp-server/rtsp-media-factory-uri.c:
1587 * gst/rtsp-server/rtsp-media.c:
1588 rtsp-media: Try to get dynamic payloaders by name from their bin first
1589 First try "pay", then "pay_%s" (where %s == pad name). And only then
1590 fall back to the code that simply takes the first payloader that is
1592 The current code usually works (but is racy) because it will always take
1593 the payloader that was last added (due to g_list_prepend() when adding
1594 elements) in pad-added and that's usually the correct one. But if a new
1595 payloader is added between pad-added and us trying to get it, we would
1596 get the wrong payloader.
1598 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1600 * tests/check/gst/client.c:
1601 client test: expect any port in transport
1602 setup_multicast_client sets a 5000-5010 range for the client
1603 ports, it is incorrect to expect the transport to always use
1607 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1609 * tests/check/gst/onvif.c:
1610 onvif tests: use g_cond_wait() correctly
1611 g_cond_wait() has to be called in a loop until required conditions
1615 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1617 * gst/rtsp-server/rtsp-stream.c:
1618 rtsp-stream: Not wait on receiver streams when pre-rolling
1619 Without this patch there are problem pre-rolling when using audio back
1621 Without this patch a probe will be created for all streams including
1622 the stream for audio backchannel. To pre-roll all this pads have to
1623 receive data. Since the stream for audio backchannel is a receiver this
1625 The solution is to never create any probes for streams that are for
1626 incomming data and instead set them as blocking already from beginning.
1628 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1630 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1631 * gst/rtsp-server/rtsp-onvif-media.c:
1632 onvif-media: fix "void function returning a value" compiler warning
1634 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1636 * gst/rtsp-server/rtsp-media.c:
1637 rtsp-media: make sure streams are blocked when sending seek
1638 The recent ONVIF work exposed a race condition when dealing with
1639 multiple streams: one of the sinks may preroll before other streams
1640 have started flushing. This led to the pipeline posting async-done
1641 prematurely, when some streams were actually still in the middle
1642 of performing a flushing seek. The newly-added code looks up a
1643 sticky segment event on the first stream in order to respond to
1644 the PLAY request with accurate Scale and Speed headers. In the
1645 failure condition, the first stream was flushing, and thus had
1646 no sticky segment event, leading to the PLAY request failing,
1647 and in turn the test.
1649 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
1652 * gst/rtsp-server/rtsp-media-factory-uri.h:
1655 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1657 * gst/rtsp-server/rtsp-client.c:
1658 * gst/rtsp-server/rtsp-client.h:
1659 * gst/rtsp-server/rtsp-media.c:
1660 * gst/rtsp-server/rtsp-media.h:
1661 * gst/rtsp-server/rtsp-onvif-client.c:
1662 * gst/rtsp-server/rtsp-onvif-client.h:
1663 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1664 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1665 * gst/rtsp-server/rtsp-onvif-media.c:
1666 * gst/rtsp-server/rtsp-onvif-server.h:
1667 * gst/rtsp-server/rtsp-stream.c:
1668 * gst/rtsp-server/rtsp-stream.h:
1669 * tests/check/gst/media.c:
1670 * tests/check/gst/onvif.c:
1671 * tests/check/meson.build:
1672 onvif: Implement and test the Streaming Specification
1673 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
1675 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1677 * gst/rtsp-server/rtsp-client.c:
1678 * gst/rtsp-server/rtsp-client.h:
1679 rtsp-client: add gst_rtsp_client_get_stream_transport()
1680 This will be used in the onvif tests in order to validate the
1681 data transmitted over TCP: for streaming to continue after a
1682 data message has been provided to client->send_func, the client
1683 is responsible for marking the message as sent on the relevant
1686 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1688 * gst/rtsp-server/rtsp-client.c:
1689 client: Scale implies TRICK_MODE
1691 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1693 * gst/rtsp-server/rtsp-client.c:
1694 client: compare booleans, not pointers to them
1696 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
1698 * gst/rtsp-server/rtsp-media.c:
1699 * gst/rtsp-server/rtsp-stream.c:
1700 * tests/check/gst/media.c:
1701 Reverse playback support
1702 GStreamer plays segment from stop to start when doing reverse playback.
1703 RTSP implies that media should be played from start of Range header to
1704 its stop. Hence we swap start and stop times before passing them to
1706 Also make gst_rtsp_stream_query_stop always return value that can be
1707 used as stop time of Range header.
1709 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
1711 * gst/rtsp-server/rtsp-client.c:
1712 * gst/rtsp-server/rtsp-media.c:
1713 * gst/rtsp-server/rtsp-media.h:
1714 * tests/check/gst/client.c:
1715 rtsp-client: add support for Scale and Speed header
1716 Add support for the RTSP Scale and Speed headers by setting the rate in
1717 the seek to (scale*speed). We then check the resulting segment for rate
1718 and applied rate, and use them as values for the Speed and Scale headers
1720 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1722 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
1724 * gst/rtsp-server/rtsp-client.c:
1725 * gst/rtsp-server/rtsp-client.h:
1726 rtsp-client: allow sub classes to adjust the seek
1727 Adds a new virtual function, adjust_play_mode(), that allows
1728 sub classes to adjust the seek done on the media. The sub class can
1729 modify the values of the the seek flags and the rate.
1730 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1732 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
1734 * gst/rtsp-server/rtsp-media.c:
1735 * gst/rtsp-server/rtsp-media.h:
1736 * gst/rtsp-server/rtsp-stream.c:
1737 * gst/rtsp-server/rtsp-stream.h:
1738 * tests/check/gst/media.c:
1739 rtsp-media: allow specifying rate when seeking
1740 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
1741 caller to specify the rate for the seek. Also added functions in
1742 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
1743 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1745 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
1749 meson: Bump minimal GLib version to 2.44
1750 This means we can use some newer features and get rid of some
1751 boilerplate code using the G_DECLARE_* macros.
1752 As discussed on IRC, 2.44 is old enough by now to start depending on it.
1754 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1756 * docs/libs/.gitignore:
1757 * docs/libs/Makefile.am:
1758 * docs/libs/gst-rtsp-server-docs.sgml:
1759 * docs/libs/gst-rtsp-server-sections.txt:
1760 * docs/libs/gst-rtsp-server.types:
1761 docs: remove obsolete gtk-doc related files
1763 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1765 * gst/rtsp-sink/gstrtspclientsink.c:
1766 doc: remove xml from comments
1768 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
1770 * docs/gst_plugins_cache.json:
1772 docs: Stop building the doc cache by default
1773 And update the cache
1774 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
1776 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
1778 * docs/gst_plugins_cache.json:
1779 docs: Update plugins documentation cache
1781 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
1784 * gst/rtsp-server/rtsp-context.c:
1785 * gst/rtsp-server/rtsp-session-pool.c:
1786 doc: Fix some docstrings
1788 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
1794 * docs/gst_plugins_cache.json:
1797 * docs/plugin-index.md:
1798 * docs/plugin-sitemap.txt:
1801 * docs/version.entities.in:
1802 * gst/rtsp-server/meson.build:
1803 * gst/rtsp-sink/meson.build:
1805 * meson_options.txt:
1806 docs: Port to hotdoc
1808 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1810 * gst/rtsp-server/rtsp-auth.c:
1811 * gst/rtsp-server/rtsp-client.h:
1812 rtsp-server: Fix various Since markers
1814 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1816 * gst/rtsp-server/rtsp-media.c:
1817 * gst/rtsp-server/rtsp-sdp.c:
1818 * gst/rtsp-server/rtsp-session-media.c:
1819 * gst/rtsp-server/rtsp-stream.c:
1820 rtsp-server: Add various Since: 1.14 markers
1822 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1824 * gst/rtsp-server/rtsp-media-factory.c:
1825 * gst/rtsp-server/rtsp-media.c:
1826 * gst/rtsp-server/rtsp-stream-transport.c:
1827 * gst/rtsp-server/rtsp-stream.c:
1828 rtsp-server: Add various missing Since: 1.16 markers
1830 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
1832 * gst/rtsp-sink/gstrtspclientsink.c:
1833 rtspclientsink: Set async-handling=false for the internal bins
1834 Without this we can easily run into a race condition with async state changes:
1835 - the pipeline is doing an async state change
1836 - we set the internal bins to PLAYING but that's ignored because an
1837 async state change is currently pending
1838 - the async state change finishes but does not change the state of the
1839 internal bins because of locked_state==TRUE
1840 - the internal bins stay in PAUSED forever
1842 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1844 * gst/rtsp-sink/gstrtspclientsink.c:
1845 rtspclientsink: Use write_messages() API to send buffer lists in one go
1846 And to write messages with multiple memories also via writev().
1848 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
1850 * gst/rtsp-server/rtsp-client.c:
1851 * gst/rtsp-server/rtsp-client.h:
1852 * gst/rtsp-server/rtsp-server-object.h:
1853 * gst/rtsp-server/rtsp-server.c:
1854 rtsp-client: Handle Content-Length limitation
1855 Add functionality to limit the Content-Length.
1856 API addition, Enhancement.
1857 Define an appropriate request size limit and reject requests
1858 exceeding the limit with response status 413 Request Entity Too Large
1861 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1868 === release 1.16.0 ===
1870 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1876 * gst-rtsp-server.doap:
1880 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
1882 * gst/rtsp-sink/gstrtspclientsink.c:
1883 rtspclientsink: Notify the stream transport about each written message
1884 Otherwise it will never try to send us the next one: it tries to keep
1885 exactly one message in-flight all the time.
1886 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
1887 in the client sink we always write data out synchronously.
1889 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
1891 * gst/rtsp-server/rtsp-stream.c:
1892 rtsp_server: Free thread pool before clean transport cache
1893 If not waiting for free thread pool before clean transport caches, there
1894 can be a crash if a thread is executing in transport list loop in
1895 function send_tcp_message.
1896 Also add a check if priv->send_pool in on_message_sent to avoid that a
1897 new thread is pushed during wait of free thread pool. This is possible
1898 since when waiting for free thread pool mutex have to be unlocked.
1900 === release 1.15.90 ===
1902 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
1908 * gst-rtsp-server.doap:
1912 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
1914 * gst/rtsp-server/rtsp-stream.c:
1915 rtsp-stream: Add support for GCM (RFC 7714)
1918 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
1920 * gst/rtsp-server/rtsp-session-pool.c:
1921 session pool: fix missing klass-> in klass->create_session
1923 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1926 g-i: pass --quiet to g-ir-scanner
1927 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
1928 that we get even if everything works just fine.
1929 We still get g-ir-scanner warnings and compiler warnings if
1930 we pass this option.
1932 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1935 g-i: silence 'nested extern' compiler warnings when building scanner binary
1936 We need a nested extern in our init section for the scanner binary
1937 so we can call gst_init to make sure GStreamer types are initialised
1938 (they are not all lazy init via get_type functions, but some are in
1939 exported variables). There doesn't seem to be any other mechanism to
1940 achieve this, so just remove that warning, it's not important at all.
1942 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1945 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
1947 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
1949 * gst/rtsp-server/rtsp-media.c:
1950 * tests/check/gst/media.c:
1951 rtsp-media: Handle set state when preparing.
1952 Handle the situation when a call to gst_rtsp_media_set_state is done
1953 when media status is preparing.
1954 Also add unit test for this scenario.
1955 The unit test simulate on a media level when two clients share a (live)
1957 Both clients have done SETUP and got responses. Now client 1 is doing
1958 play and client 2 is just closing the connection.
1959 Then without patch there are a problem when
1960 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
1961 And client2 is doing closing connection we can end up in a call
1962 to gst_rtsp_media_set_state when
1963 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
1964 shut down media is jumped over .
1965 With this patch and this scenario we wait until
1966 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
1967 execute after that and now we will execute the logic for
1970 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
1978 === release 1.15.2 ===
1980 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
1986 * gst-rtsp-server.doap:
1990 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
1992 * gst/rtsp-server/rtsp-media.c:
1993 * tests/check/gst/client.c:
1994 rtsp-media: Fix multicast use case with common media
2003 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2005 * gst/rtsp-server/rtsp-client.c:
2006 * gst/rtsp-server/rtsp-stream.c:
2007 * gst/rtsp-server/rtsp-stream.h:
2008 rtsp-server: remove recursive behavior
2009 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2011 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2013 * gst/rtsp-server/rtsp-client.c:
2014 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2015 And route all messages through the send_func if no send_messages_func
2017 We otherwise break backwards compatibility.
2019 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2021 * docs/libs/gst-rtsp-server-sections.txt:
2022 * gst/rtsp-server/rtsp-client.c:
2023 * gst/rtsp-server/rtsp-client.h:
2024 * gst/rtsp-server/rtsp-stream.c:
2025 rtsp-client: Add support for sending buffer lists directly
2026 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2028 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2030 * docs/libs/gst-rtsp-server-sections.txt:
2031 * gst/rtsp-server/rtsp-client.c:
2032 * gst/rtsp-server/rtsp-media.c:
2033 * gst/rtsp-server/rtsp-stream-transport.c:
2034 * gst/rtsp-server/rtsp-stream-transport.h:
2035 * gst/rtsp-server/rtsp-stream.c:
2036 * gst/rtsp-sink/gstrtspclientsink.c:
2037 rtsp-server: Add support for buffer lists
2038 This adds new functions for passing buffer lists through the different
2039 layers without breaking API/ABI, and enables the appsink to actually
2040 provide buffer lists.
2041 This should already reduce CPU usage and potentially context switches a
2042 bit by passing a whole buffer list from the appsink instead of
2043 individual buffers. As a next step it would be necessary to
2044 a) Add support for a vector of data for the GstRTSPMessage body
2045 b) Add support for sending multiple messages at once to the
2046 GstRTSPWatch and let it be handled internally
2047 c) Adding API to GOutputStream that works like writev()
2048 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2050 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2052 * gst/rtsp-server/rtsp-client.c:
2053 client: Fix crash in close handler
2054 The close handler could trigger a crash because it invalidated the
2055 watch_context while still leaving a source attached to it which would be
2056 cleaned up at a later point.
2058 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2060 * gst/rtsp-server/rtsp-stream.c:
2061 rtsp-stream: Use cached address when allocating sockets
2062 If an address/port was previously decided upon (ex: multicast in the
2063 SDP), then use that instead of re-creating another one
2064 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2066 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2068 * gst/rtsp-server/rtsp-media.c:
2069 rtsp-media: Fix race codition in finish_unprepare
2070 The previous fix for race condition around finish_unprepare where the
2071 function could be called twice assumed that the status wouldn't change
2072 during execution of the function. This assumption is incorrect as the
2073 state may change, for example if an error message arrives from the
2075 Instead a flag keeping track on whether the finish_unprepare function
2076 is currently executing is introduced and checked.
2077 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2079 === release 1.15.1 ===
2081 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2087 * gst-rtsp-server.doap:
2091 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2093 * gst/rtsp-server/rtsp-stream.c:
2094 Add source elements to the pipeline before activation
2095 In plug_src we changed the element state before adding it to
2096 the owner container. This prevented the pipeline from intercepting
2097 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2098 to assign a custom task pool.
2099 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2101 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2104 Automatic update of common submodule
2105 From ed78bee to 59cb678
2107 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2109 * examples/test-appsrc.c:
2110 examples: test-appsrc: fix coding style error
2112 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2114 * examples/test-appsrc.c:
2115 examples: test-appsrc: fix buffer leak
2117 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2119 * gst/rtsp-server/rtsp-media.c:
2120 rtsp-media: Update priv->blocked when linked streams are unblocked.
2121 Media is considered to be blocked when all streams that belong to
2122 that media are blocked.
2123 This patch solves the problem of inconsistent updates of
2124 priv->blocked that are not synchronized with the media state.
2126 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2128 * gst/rtsp-server/rtsp-media.c:
2129 rtsp-media: Don't block streams before seeking
2130 Before the seek operation is performed on media, it's required that
2131 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2132 At this stage, all transport parts (transport sinks) have been successfully
2133 added to the pipeline and there is no need for blocking the streams.
2135 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2137 * tests/check/gst/rtspserver.c:
2138 tests: rtspserver: Add shared media test case for TCP
2140 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2142 * gst/rtsp-server/rtsp-stream.c:
2143 rtsp-stream: Use seqnum-offset for rtpinfo
2144 The sequence number in the rtpinfo is supposed to be the first RTP
2145 sequence number. The "seqnum" property on a payloader is supposed to be
2146 the number from the last processed RTP packet. The sequence number for
2147 payloaders that inherit gstrtpbasepayload will not be correct in case of
2148 buffer lists. In order to fix the seqnum property on the payloaders
2149 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2150 "seqnum-offset" from the "stats" property contains the value of the
2151 very first RTP packet in a stream. The server will, however, try to look
2152 at the last simple in the sink element and only use properties on the
2153 payloader in case there no sink elements yet, and by looking at the last
2154 sample of the sink gives the server full control of which RTP packet it
2155 looks at. If the payloader does not have the "stats" property, "seqnum"
2156 is still used since "seqnum-offset" is only present in as part of
2157 "stats" and this is still an issue not solved with this patch.
2158 Needed for gst-plugins-base!17
2160 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2162 * gst/rtsp-server/rtsp-stream.c:
2163 rtsp-stream: Plug memory leak
2164 Attaching a GSource to a context will increase the refcount. The idle
2165 source will never be free'd since the initial reference is never
2168 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2171 Add Gitlab CI configuration
2172 This commit adds a .gitlab-ci.yml file, which uses a feature
2173 to fetch the config from a centralized repository. The intent is
2174 to have all the gstreamer modules use the same configuration.
2175 The configuration is currently hosted at the gst-ci repository
2176 under the gitlab/ci_template.yml path.
2177 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2179 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2182 * gst-rtsp-server.doap:
2183 Update git locations to gitlab
2185 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2187 * gst/rtsp-server/meson.build:
2188 meson: add new onvif types
2190 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2192 * gst/rtsp-server/meson.build:
2193 Add ONVIF subclass headers to the installed headers in meson.build too
2195 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2197 * gst/rtsp-server/rtsp-server-object.h:
2198 * gst/rtsp-server/rtsp-server.h:
2199 rtsp-server: Declare GstRTSPServer struct before anything else
2200 It's needed by all kinds of other headers, including the ones that are
2201 required for defining the GstRTSPServer struct itself and its API.
2203 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2205 * gst/rtsp-server/rtsp-onvif-client.h:
2206 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2207 * gst/rtsp-server/rtsp-onvif-media.h:
2208 * gst/rtsp-server/rtsp-onvif-server.h:
2209 Mark all ONVIF-specific subclasses as Since 1.14
2211 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2213 * gst/rtsp-server/Makefile.am:
2214 * gst/rtsp-server/meson.build:
2215 * gst/rtsp-server/rtsp-context.h:
2216 * gst/rtsp-server/rtsp-onvif-server.c:
2217 * gst/rtsp-server/rtsp-onvif-server.h:
2218 * gst/rtsp-server/rtsp-server-object.h:
2219 * gst/rtsp-server/rtsp-server-prelude.h:
2220 * gst/rtsp-server/rtsp-server.c:
2221 * gst/rtsp-server/rtsp-server.h:
2222 * gst/rtsp-server/rtsp-session.h:
2223 Include ONVIF types from single-include rtsp-server.h
2224 ... by actually making it a single-include header and moving everything
2225 related to the GstRTSPServer type to rtsp-server-object.h instead.
2226 Otherwise there are too many circular includes.
2227 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2229 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2231 * gst/rtsp-server/rtsp-client.c:
2232 * gst/rtsp-server/rtsp-latency-bin.c:
2233 * gst/rtsp-server/rtsp-stream.c:
2234 * gst/rtsp-server/rtsp-stream.h:
2235 rtsp-stream: use idle source in on_message_sent
2236 When the underlying layers are running on_message_sent, this sometimes
2237 causes the underlying layer to send more data, which will cause the
2238 underlying layer to run callback on_message_sent again. This can go on
2240 To break this chain, we introduce an idle source that takes care of
2241 sending data if there are more to send when running callback
2242 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2244 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2246 * gst/rtsp-server/rtsp-client.c:
2247 rtsp-client: Remove timeout GSource on cleanup
2248 Avoids ending up with races where a timeout would still be around
2249 *after* a client was gone. This could happen rather easily in
2250 RTSP-over-HTTP mode on a local connection, where each RTSP message
2251 would be sent as a different HTTP connection with the same tunnelid.
2252 If not properly removed, that timeout would then try to free again
2253 a client (and its contents).
2255 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2257 * gst/rtsp-server/Makefile.am:
2258 autotools: fix distcheck
2260 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2262 * gst/rtsp-server/Makefile.am:
2263 * gst/rtsp-server/meson.build:
2264 * gst/rtsp-server/rtsp-latency-bin.c:
2265 * gst/rtsp-server/rtsp-latency-bin.h:
2266 * gst/rtsp-server/rtsp-onvif-media.c:
2267 onvif: encapsulate onvif part into a bin
2268 ...and thus do not let onvif affect pipelines latency
2269 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2271 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2273 * tests/check/gst/client.c:
2274 tests: client: Avoid bind() failures in tests
2275 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2277 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2279 * gst/rtsp-server/rtsp-media-factory.c:
2280 * gst/rtsp-server/rtsp-media-factory.h:
2281 * gst/rtsp-server/rtsp-media.c:
2282 * gst/rtsp-server/rtsp-media.h:
2283 * gst/rtsp-server/rtsp-stream.c:
2284 * gst/rtsp-server/rtsp-stream.h:
2285 * tests/check/gst/client.c:
2286 * tests/check/gst/mediafactory.c:
2287 New property for socket binding to mcast addresses
2288 By default the multicast sockets are bound to INADDR_ANY,
2289 as it's not allowed to bind sockets to multicast addresses
2290 in Windows. This default behaviour can be changed by setting
2291 bind-mcast-address property on the media-factory object.
2292 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2294 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2297 * gst/rtsp-server/Makefile.am:
2298 * gst/rtsp-server/meson.build:
2299 * gst/rtsp-server/rtsp-address-pool.c:
2300 * gst/rtsp-server/rtsp-auth.c:
2301 * gst/rtsp-server/rtsp-client.c:
2302 * gst/rtsp-server/rtsp-context.c:
2303 * gst/rtsp-server/rtsp-media-factory-uri.c:
2304 * gst/rtsp-server/rtsp-media-factory.c:
2305 * gst/rtsp-server/rtsp-media.c:
2306 * gst/rtsp-server/rtsp-mount-points.c:
2307 * gst/rtsp-server/rtsp-params.c:
2308 * gst/rtsp-server/rtsp-permissions.c:
2309 * gst/rtsp-server/rtsp-sdp.c:
2310 * gst/rtsp-server/rtsp-server-prelude.h:
2311 * gst/rtsp-server/rtsp-server.c:
2312 * gst/rtsp-server/rtsp-session-media.c:
2313 * gst/rtsp-server/rtsp-session-pool.c:
2314 * gst/rtsp-server/rtsp-session.c:
2315 * gst/rtsp-server/rtsp-stream-transport.c:
2316 * gst/rtsp-server/rtsp-stream.c:
2317 * gst/rtsp-server/rtsp-thread-pool.c:
2318 * gst/rtsp-server/rtsp-token.c:
2320 libs: fix API export/import and 'inconsistent linkage' on MSVC
2321 Export rtsp-server library API in headers when we're building the
2322 library itself, otherwise import the API from the headers.
2323 This fixes linker warnings on Windows when building with MSVC.
2324 Fix up some missing config.h includes when building the lib which
2325 is needed to get the export api define from config.h
2326 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2328 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2330 * gst/rtsp-server/rtsp-media-factory.c:
2331 rtsp-media-factory: Add missing break statements
2332 This resulted in warnings/assertions whenever one accessed the
2333 max-mcast-ttl property.
2337 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2340 * meson_options.txt:
2341 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2343 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2346 * meson_options.txt:
2347 * tests/check/meson.build:
2348 meson: add option to disable build of rtspclientsink plugin
2350 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2352 * meson_options.txt:
2353 meson: re-arrange options
2355 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2358 * meson_options.txt:
2359 * tests/check/meson.build:
2360 * tests/meson.build:
2361 meson: Use feature option for tests option
2362 This was somehow missed the last time around.
2364 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2366 * gst/rtsp-server/meson.build:
2368 meson: Maintain macOS ABI through dylib versioning
2369 Requires Meson 0.48, but the feature will be ignored on older versions
2370 so it's safe to add it without bumping the requirement.
2372 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2374 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2376 * gst/rtsp-sink/meson.build:
2378 meson: add pkg-config file for the rtspclientsink plugin
2380 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2382 * gst/rtsp-server/rtsp-client.c:
2383 * tests/check/gst/client.c:
2384 rtsp-client: Avoid reuse of channel numbers for interleaved
2385 If a (strange) client would reuse interleaved channel numbers in
2386 multiple SETUP requests, we should not accept them. The channel
2387 numbers are used for looking up stream transports in the
2388 priv->transports hash table, and transports disappear from the table
2389 if channel numbers are reused.
2390 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2391 server to change the channel numbers suggested by the client.
2392 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2394 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2396 * tests/check/gst/client.c:
2397 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2398 Allow regex for matching transport header against expected pattern.
2399 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2401 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2403 * tests/check/meson.build:
2404 meson: There is no gstreamer-plugins-good-1.0.pc
2405 There is no installed version of that, only an uninstalled version.
2407 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2409 * gst/rtsp-server/rtsp-client.c:
2410 * tests/check/gst/stream.c:
2411 Fix indentation again
2413 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2415 * gst/rtsp-server/rtsp-client.c:
2416 * gst/rtsp-server/rtsp-stream.c:
2417 * gst/rtsp-server/rtsp-stream.h:
2418 * tests/check/gst/client.c:
2419 * tests/check/gst/stream.c:
2420 stream: Added a list of multicast client addresses
2421 When media is shared, the same media stream can be sent
2422 to multiple multicast groups. Currently, there is no API
2423 to retrieve multicast addresses from the stream.
2424 When calling gst_rtsp_stream_get_multicast_address() function,
2425 only the first multicast address is returned.
2426 With this patch, each multicast destination requested in SETUP
2427 will be stored in an internal list (call to
2428 gst_rtsp_stream_add_multicast_client_address()).
2429 The list of multicast groups requested by the clients can be
2430 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2431 There still exist some problems with the current implementation
2432 in the multicast case:
2433 1) The receiving part is currently only configured with
2434 regard to the first multicast client (see
2435 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2436 2) Secondly, of security reasons, some constraints should be
2437 put on the requested multicast destinations (see
2438 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2439 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2440 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2442 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2444 * gst/rtsp-server/rtsp-client.c:
2445 * gst/rtsp-server/rtsp-stream.c:
2446 * gst/rtsp-server/rtsp-stream.h:
2447 * tests/check/gst/client.c:
2448 stream: Choose the maximum ttl value provided by multicast clients
2449 The maximum ttl value provided so far by the multicast clients
2450 will be chosen and reported in the response to the current
2452 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2453 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2455 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2457 * gst/rtsp-server/rtsp-stream.c:
2458 * tests/check/gst/client.c:
2459 rtsp-stream: Don't require address pool in the transport specific case
2460 If "transport.client-settings" parameter is set to true, the client is
2461 allowed to specify destination, ports and ttl.
2462 There is no need for pre-configured address pool.
2463 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2464 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2466 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2468 * gst/rtsp-server/rtsp-client.c:
2469 * tests/check/gst/client.c:
2470 client: Don't reserve multicast address in the client setting case
2471 When two multicast clients request specific transport
2472 configurations, and "transport.client-settings" parameter is
2473 set to true, it's wrong to actually require that these two
2474 clients request the same multicast group.
2475 Removed test_client_multicast_invalid_transport_specific test
2476 cases as they wrongly require that the requested destination
2477 address is supposed to be present in the address pool, also in
2478 the case when "transport.client-settings" parameter is set to true.
2479 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2480 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2482 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2484 * gst/rtsp-server/rtsp-media-factory.c:
2485 * gst/rtsp-server/rtsp-media-factory.h:
2486 * gst/rtsp-server/rtsp-media.c:
2487 * gst/rtsp-server/rtsp-media.h:
2488 * gst/rtsp-server/rtsp-stream.c:
2489 * gst/rtsp-server/rtsp-stream.h:
2490 * tests/check/gst/mediafactory.c:
2491 Add new API for setting/getting maximum multicast ttl value
2492 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2493 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2495 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2497 * gst/rtsp-server/rtsp-stream.c:
2498 rtsp-stream: avoid duplicating the first multicast client
2499 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2500 clients were dynamically added and removed to the multicast
2501 udp sinks, as such we should no longer add a first client in
2502 set_multicast_socket_for_udpsink
2503 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2505 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2507 * gst/rtsp-server/rtsp-stream.c:
2508 Revert "rtsp-stream: avoid duplicating the first multicast client"
2509 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2510 Commits where accidentially squashed together
2512 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2514 * gst/rtsp-server/rtsp-client.c:
2515 * gst/rtsp-server/rtsp-media-factory.c:
2516 * gst/rtsp-server/rtsp-media-factory.h:
2517 * gst/rtsp-server/rtsp-media.c:
2518 * gst/rtsp-server/rtsp-media.h:
2519 * gst/rtsp-server/rtsp-stream.c:
2520 * gst/rtsp-server/rtsp-stream.h:
2521 * tests/check/gst/client.c:
2522 * tests/check/gst/mediafactory.c:
2523 Revert "Add new API for setting/getting maximum multicast ttl value"
2524 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2525 Commits where accidentially squashed together
2527 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2529 * gst/rtsp-server/rtsp-stream.c:
2530 * tests/check/gst/client.c:
2531 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2532 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2533 Commits where accidentially squashed together
2535 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.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 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2543 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2544 Commits where accidentially squashed together
2546 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2548 * examples/test-auth-digest.c:
2549 examples: Fix indentation
2551 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2553 * gst/rtsp-server/rtsp-client.c:
2554 * gst/rtsp-server/rtsp-stream.c:
2555 * gst/rtsp-server/rtsp-stream.h:
2556 * tests/check/gst/client.c:
2557 * tests/check/gst/stream.c:
2558 stream: Choose the maximum ttl value provided by multicast clients
2559 The maximum ttl value provided so far by the multicast clients
2560 will be chosen and reported in the response to the current
2562 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2564 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2566 * gst/rtsp-server/rtsp-stream.c:
2567 * tests/check/gst/client.c:
2568 rtsp-stream: Don't require address pool in the transport specific case
2569 If "transport.client-settings" parameter is set to true, the client is
2570 allowed to specify destination, ports and ttl.
2571 There is no need for pre-configured address pool.
2572 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2574 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2576 * gst/rtsp-server/rtsp-client.c:
2577 * gst/rtsp-server/rtsp-media-factory.c:
2578 * gst/rtsp-server/rtsp-media-factory.h:
2579 * gst/rtsp-server/rtsp-media.c:
2580 * gst/rtsp-server/rtsp-media.h:
2581 * gst/rtsp-server/rtsp-stream.c:
2582 * gst/rtsp-server/rtsp-stream.h:
2583 * tests/check/gst/client.c:
2584 * tests/check/gst/mediafactory.c:
2585 Add new API for setting/getting maximum multicast ttl value
2586 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2588 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2590 * gst/rtsp-server/rtsp-stream.c:
2591 rtsp-stream: avoid duplicating the first multicast client
2592 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2593 clients were dynamically added and removed to the multicast
2594 udp sinks, as such we should no longer add a first client in
2595 set_multicast_socket_for_udpsink
2596 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2598 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2600 * gst/rtsp-server/Makefile.am:
2601 rtsp-server: Add gstreamer-base gir dir in autotools
2603 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2605 * gst/rtsp-server/rtsp-client.c:
2606 * gst/rtsp-server/rtsp-stream.c:
2607 rtsp-client: always allocate both IPV4 and IPV6 sockets
2608 multiudpsink does not support setting the socket* properties
2609 after it has started, which meant that rtsp-server could no
2610 longer serve on both IPV4 and IPV6 sockets since the patches
2611 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2613 When first connecting an IPV6 client then an IPV4 client,
2614 multiudpsink fell back to using the IPV6 socket.
2615 When first connecting an IPV4 client, then an IPV6 client,
2616 multiudpsink errored out, released the IPV4 socket, then
2617 crashed when trying to send a message on NULL nevertheless,
2618 that is however a separate issue.
2619 This could probably be fixed by handling the setting of
2620 sockets in multiudpsink after it has started, that will
2621 however be a much more significant effort.
2622 For now, this commit simply partially reverts the behaviour
2623 of rtsp-stream: it will continue to only create the udpsinks
2624 when needed, as was the case since the patches were merged,
2625 it will however when creating them, always allocate both
2626 sockets and set them on the sink before it starts, as was
2627 the case prior to the patches.
2628 Transport configuration will only error out if the allocation
2629 of UDP sockets fails for the actual client's family, this
2630 also downgrades the GST_ERRORs in alloc_ports_one_family
2631 to GST_WARNINGs, as failing to allocate is no longer
2633 https://bugzilla.gnome.org/show_bug.cgi?id=796875
2635 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2638 * meson_options.txt:
2639 meson: Convert common options to feature options
2640 These are necessary for gst-build to set options correctly. The
2641 remaining automagic option is cgroup support in examples.
2642 https://bugzilla.gnome.org/show_bug.cgi?id=795107
2644 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2646 * gst/rtsp-server/rtsp-stream.c:
2647 rtsp-stream: Slightly simplify locking
2649 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
2651 * gst/rtsp-server/rtsp-client.c:
2652 * gst/rtsp-server/rtsp-stream-transport.c:
2653 * gst/rtsp-server/rtsp-stream-transport.h:
2654 * gst/rtsp-server/rtsp-stream.c:
2655 Limit queued TCP data messages to one per stream
2656 Before, the watch backlog size in GstRTSPClient was changed
2657 dynamically between unlimited and a fixed size, trying to avoid both
2658 unlimited memory usage and deadlocks while waiting for place in the
2659 queue. (Some of the deadlocks were described in a long comment in
2661 In the previous commit, we changed to a fixed backlog size of 100.
2662 This is possible, because we now handle RTP/RTCP data messages differently
2663 from RTSP request/response messages.
2664 The data messages are messages tunneled over TCP. We allow at most one
2665 queued data message per stream in GstRTSPClient at a time, and
2666 successfully sent data messages are acked by sending a "message-sent"
2667 callback from the GstStreamTransport. Until that ack comes, the
2668 GstRTSPStream does not call pull_sample() on its appsink, and
2669 therefore the streaming thread in the pipeline will not be blocked
2670 inside GstRTSPClient, waiting for a place in the queue.
2671 pull_sample() is called when we have both an ack and a "new-sample"
2672 signal from the appsink. Then, we know there is a buffer to write.
2673 RTSP request/response messages are not acked in the same way as data
2674 messages. The rest of the 100 places in the queue are used for
2675 them. If the queue becomes full of request/response messages, we
2676 return an error and close the connection to the client.
2677 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2679 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
2681 * gst/rtsp-server/rtsp-client.c:
2682 rtsp-client: Use fixed backlog size
2683 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
2684 Preparation for the next commit, which changes to a different way of
2685 avoiding both deadlocks and unlimited memory usage with the watch
2688 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2690 * gst/rtsp-server/rtsp-media.c:
2691 rtsp-media: unref clock (if set) when finalizing
2692 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2694 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2696 * docs/libs/gst-rtsp-server-sections.txt:
2697 rtsp-media: add gst_rtsp_media_*_set_clock to docs
2698 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2700 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2702 * gst/rtsp-server/rtsp-media-factory.c:
2703 media-factory: unref old clock when setting new clock
2704 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2706 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
2708 * gst/rtsp-server/rtsp-media-factory.c:
2709 media-factory: unref clock in finalize
2710 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2712 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2714 * gst/rtsp-server/rtsp-onvif-media.c:
2715 rtsp-onvif-media: fix g-ir-scanner warnings
2717 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2720 .gitignore: add another example binary
2722 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
2724 * examples/meson.build:
2725 meson: add new test-appsrc2 example to meson build
2727 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2729 * examples/Makefile.am:
2730 examples: fix build of new test-appsrc2 example
2731 Need to link against libgstapp-1.0.
2733 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
2735 * examples/.gitignore:
2736 * examples/Makefile.am:
2737 * examples/test-appsrc2.c:
2738 examples: Add test-appsrc2
2739 Add an example of feeding both audio and video into an RTSP
2740 pipeline via appsrc.
2742 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
2744 * gst/rtsp-server/rtsp-client.c:
2745 client: Strip transport parts as whitespaces could be around commas
2746 https://bugzilla.gnome.org/show_bug.cgi?id=758428
2748 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
2750 * gst/rtsp-server/rtsp-stream.c:
2751 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
2752 Fix race when setting up source elements.
2753 Since we set the source element(s) to PLAYING state before hooking
2754 them up to the downstream funnel, it's possible for the source element
2755 to receive packets before we actually get to linking it to the funnel,
2756 in which case buffers would be pushed out on an unlinked pad, causing
2757 it to error out and stop receiving more data.
2758 We fix this by blocking the source's srcpad until we have linked it.
2759 https://bugzilla.gnome.org/show_bug.cgi?id=796160
2761 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
2763 * gst/rtsp-server/rtsp-stream.c:
2764 rtsp-stream: Fix mismatch between allowed and configured protocols
2765 https://bugzilla.gnome.org/show_bug.cgi?id=796679
2767 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
2769 * gst/rtsp-server/rtsp-stream.c:
2770 rtsp-stream: Emit a signal when the SRTP decoder is created
2771 https://bugzilla.gnome.org/show_bug.cgi?id=778080
2773 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
2775 * gst/rtsp-server/rtsp-stream.c:
2776 rtsp-stream: Don't require presence of sinks in _get_*_socket()
2777 Transport specific sink elements are added to the pipeline
2778 in PLAY request and sockets are already created in SETUP so
2779 it's actually wrong to require the presence of sinks in
2780 _get_*_socket() functions.
2781 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2783 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
2785 * gst/rtsp-server/rtsp-stream.c:
2786 rtsp-stream: Update transport for multicast clients as well
2787 If a multicast client requests different transport settings
2788 than the existing one make sure that this new transport
2789 configuruation is propagated to the multicast udp sink.
2790 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2792 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
2794 * gst/rtsp-server/rtsp-stream.c:
2795 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
2796 And not on unicast udp sinks
2797 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2799 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
2801 * gst/rtsp-server/rtsp-address-pool.c:
2802 * gst/rtsp-server/rtsp-auth.c:
2803 * gst/rtsp-server/rtsp-client.c:
2804 * gst/rtsp-server/rtsp-media-factory-uri.c:
2805 * gst/rtsp-server/rtsp-media-factory.c:
2806 * gst/rtsp-server/rtsp-media.c:
2807 * gst/rtsp-server/rtsp-mount-points.c:
2808 * gst/rtsp-server/rtsp-server.c:
2809 * gst/rtsp-server/rtsp-session-media.c:
2810 * gst/rtsp-server/rtsp-session-pool.c:
2811 * gst/rtsp-server/rtsp-session.c:
2812 * gst/rtsp-server/rtsp-stream-transport.c:
2813 * gst/rtsp-server/rtsp-stream.c:
2814 * gst/rtsp-server/rtsp-thread-pool.c:
2815 Update for g_type_class_add_private() deprecation in recent GLib
2817 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
2819 * gst/rtsp-server/rtsp-auth.c:
2820 * gst/rtsp-server/rtsp-media.c:
2821 * gst/rtsp-server/rtsp-sdp.c:
2822 * gst/rtsp-server/rtsp-stream.c:
2825 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
2827 * examples/Makefile.am:
2828 * examples/test-video-disconnect.c:
2829 examples: Add test-video-disconnect example
2830 Simple example which cuts off all clients 10 seconds
2831 after the first one connects.
2833 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2835 * docs/libs/gst-rtsp-server-sections.txt:
2836 * examples/test-auth-digest.c:
2837 * gst/rtsp-server/rtsp-auth.c:
2838 * gst/rtsp-server/rtsp-auth.h:
2839 rtsp-auth: Add support for parsing .htdigest files
2840 Passwords are usually not stored in clear text, but instead
2841 stored already hashed in a .htdigest file.
2842 Add support for parsing such files, add API to allow setting
2843 a custom realm in RTSPAuth, and update the digest example.
2844 https://bugzilla.gnome.org/show_bug.cgi?id=796637
2846 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
2848 * gst/rtsp-sink/gstrtspclientsink.c:
2849 * gst/rtsp-sink/gstrtspclientsink.h:
2850 rtspclientsink: fix waiting for multiple streams
2851 We were previously only ever waiting for a single stream to notify it's
2852 blocked status through GstRTSPStreamBlocking. Actually count streams to
2854 Fixes rtspclientsink sending SDP's without out some of the input
2856 https://bugzilla.gnome.org/show_bug.cgi?id=796624
2858 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2860 * docs/libs/gst-rtsp-server-sections.txt:
2861 docs: add missing auth methods
2863 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2865 * gst/rtsp-server/rtsp-stream.c:
2866 rtsp-stream: only create funnel if it didn't exist already.
2867 This precented using multiple protocols for the same stream.
2868 https://bugzilla.gnome.org/show_bug.cgi?id=796634
2870 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2872 * examples/meson.build:
2873 meson: build auth-digest example
2875 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
2877 * gst/rtsp-server/rtsp-client.c:
2878 * gst/rtsp-server/rtsp-media.c:
2879 * gst/rtsp-server/rtsp-sdp.c:
2880 * gst/rtsp-server/rtsp-session-media.c:
2881 * gst/rtsp-server/rtsp-stream-transport.c:
2882 Get payloader stats only for the sending streams
2883 Get/set payloader properties only for streams that actually
2884 contain a payloader element.
2885 https://bugzilla.gnome.org/show_bug.cgi?id=796523
2887 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
2889 * gst/rtsp-server/Makefile.am:
2890 Makefile: Don't hardcode libtool for g-i build
2891 Similar to the other commits in core/base/bad
2893 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
2895 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2896 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
2897 https://bugzilla.gnome.org/show_bug.cgi?id=796229
2899 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
2901 * gst/rtsp-sink/gstrtspclientsink.c:
2902 rtspclientsink: Don't deadlock in preroll on early close
2903 If the connection is closed very early, the flushing
2904 marker might not get set and rtspclientsink can get
2905 deadlocked waiting for preroll forever.
2906 https://bugzilla.gnome.org/show_bug.cgi?id=786961
2908 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2911 * meson_options.txt:
2912 meson: Update option names to omit disable_ and with- prefixes
2913 Also yield common options to the outer project (gst-build in our case)
2914 so that they don't have to be set manually.
2916 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2919 meson: use -Wl,-Bsymbolic-functions where supported
2920 Just like the autotools build.
2922 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2925 * tests/check/Makefile.am:
2926 configure: check for -good and -bad plugins only in uninstalled setup
2927 Avoids confusing configure messages looking or a -good .pc file
2929 Also use plugindir variables that common macros set while at it.
2930 https://bugzilla.gnome.org/show_bug.cgi?id=795466
2932 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
2934 * gst/rtsp-server/rtsp-client.c:
2935 rtsp-client: Fix session timeout
2936 When streaming data over TCP then is not the keep-alive
2937 functionality working.
2938 The reason is that the function do_send_data have changed
2939 to boolean but the code is still checking the received result
2940 from send_func with GST_RTSP_OK.
2941 The result is that a successful send_func will always lead to
2942 that do_send_data is returning false and the keep-alive will
2944 https://bugzilla.gnome.org/show_bug.cgi?id=795321
2946 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2948 * docs/libs/gst-rtsp-server-sections.txt:
2949 * gst/rtsp-server/rtsp-media.c:
2950 * gst/rtsp-server/rtsp-sdp.c:
2951 * gst/rtsp-server/rtsp-stream.c:
2952 * gst/rtsp-server/rtsp-stream.h:
2953 * gst/rtsp-sink/gstrtspclientsink.c:
2954 * gst/rtsp-sink/gstrtspclientsink.h:
2955 Implement support for ULP Forward Error Correction
2956 In this initial commit, interface is only exposed for RECORD,
2957 further work will be needed in rtspsrc to support this for
2959 https://bugzilla.gnome.org/show_bug.cgi?id=794911
2961 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2963 * gst/rtsp-server/rtsp-onvif-media.c:
2964 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
2965 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
2966 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
2967 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
2968 the opposite, just like the ONVIF standard.
2969 Let's follow those RFCs as we're doing RTSP here, and add a property at
2970 a later time if needed to switch to the SDP RFC behaviour.
2971 https://bugzilla.gnome.org/show_bug.cgi?id=793964
2973 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2976 Automatic update of common submodule
2977 From 3fa2c9e to ed78bee
2979 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
2981 * gst/rtsp-server/rtsp-client.c:
2982 * gst/rtsp-server/rtsp-media-factory.c:
2983 * gst/rtsp-server/rtsp-media.c:
2984 * gst/rtsp-server/rtsp-stream.c:
2985 * tests/check/gst/rtspclientsink.c:
2986 gst: Run everything through gst-indent again
2988 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
2990 * gst/rtsp-server/rtsp-media.c:
2991 * tests/check/gst/media.c:
2992 rtsp-media: query the position on active streams if media is complete
2993 If the media is complete, i.e. one or more streams have been configured
2994 with sinks, then we want to query the position on those streams only.
2995 A query on an incomplete stream may return a position that originates from
2997 https://bugzilla.gnome.org/show_bug.cgi?id=794964
2999 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3001 * gst/rtsp-sink/gstrtspclientsink.c:
3002 rtspclientsink: make sure not to use freed string
3003 Set transport string to NULL after freeing it, so that
3004 at worst we get a NULL pointer if constructing a new
3005 transport string fails (which shouldn't really fail here).
3006 Also check return value of that, just in case.
3009 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3011 * gst/rtsp-server/rtsp-client.c:
3012 rtsp-client: do not free string passed to take_header
3014 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3016 * gst/rtsp-server/rtsp-stream.c:
3017 rtsp-stream: do not take lock in request_aux_receiver
3018 Added it right before pushing the previous commit, it is
3019 incorrect and deadlocks because this function gets called
3020 from the join_bin thread, which already holds the lock,
3021 that's the reason why request_aux_sender didn't take the
3024 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3026 * docs/libs/gst-rtsp-server-sections.txt:
3027 * gst/rtsp-server/rtsp-media-factory.c:
3028 * gst/rtsp-server/rtsp-media-factory.h:
3029 * gst/rtsp-server/rtsp-media.c:
3030 * gst/rtsp-server/rtsp-media.h:
3031 * gst/rtsp-server/rtsp-stream.c:
3032 * gst/rtsp-server/rtsp-stream.h:
3033 rtsp-server: add API to enable retransmission requests
3034 "do-retransmission" was previously set when rtx-time != 0,
3035 which made no sense as do-retransmission is used to enable
3036 the sending of retransmission requests, where as rtx-time
3037 is used by the peer to enable storing of buffers in order
3038 to respond to retransmission requests.
3039 rtsp-media now also provides a callback for the
3040 request-aux-receiver signal.
3041 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3043 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3045 * gst/rtsp-sink/gstrtspclientsink.c:
3046 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3047 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3049 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3051 * gst/rtsp-sink/gstrtspclientsink.c:
3052 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3053 This in order to be able to decrypt the RTCP backchannel
3054 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3056 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3058 * gst/rtsp-server/rtsp-client.c:
3059 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3060 When sending back an encrypted RTCP back channel, it is useful
3061 for the client to know the encryption key.
3062 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3064 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3066 * gst/rtsp-server/rtsp-client.c:
3067 * gst/rtsp-server/rtsp-stream.c:
3068 * gst/rtsp-server/rtsp-stream.h:
3069 rtsp-stream: extract handle_keymgmt from rtsp-client
3070 rtspclientsink will also need to parse KeyMgmt headers
3071 sent by the server to decrypt the RTCP backchannel stream
3072 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3074 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3076 * gst/rtsp-sink/gstrtspclientsink.c:
3077 * tests/check/gst/rtspclientsink.c:
3078 rtspclientsink: Fix client ports for the RTCP backchannel
3079 This was broken since the work for delayed transport creation
3080 was merged: the creation of the transports string depends on
3081 calling stream_get_server_port, which only starts returning
3082 something meaningful after a call to stream_allocate_udp_sockets
3083 has been made, this function expects a transport that we parse
3084 from the transport string ...
3085 Significant refactoring is in order, but does not look entirely
3086 trivial, for now we put a band aid on and create a second transport
3087 string after the stream has been completed, to pass it in
3088 the request headers instead of the previous, incomplete one.
3089 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3091 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3093 * gst/rtsp-server/rtsp-client.c:
3094 rtsp-client:Error handling when equal http session cookie
3095 There are some clients that are sending same session cookie on random
3097 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3099 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3101 * gst/rtsp-server/rtsp-media-factory-uri.c:
3102 rtsp-media-factory-uri: Fix compilation with latest GLib
3103 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3104 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3105 data->factory = g_object_ref (factory);
3108 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3116 === release 1.14.0 ===
3118 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3124 * gst-rtsp-server.doap:
3128 === release 1.13.91 ===
3130 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3136 * gst-rtsp-server.doap:
3140 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3142 * gst/rtsp-server/Makefile.am:
3143 * gst/rtsp-server/meson.build:
3144 * gst/rtsp-server/rtsp-address-pool.h:
3145 * gst/rtsp-server/rtsp-auth.h:
3146 * gst/rtsp-server/rtsp-client.h:
3147 * gst/rtsp-server/rtsp-context.h:
3148 * gst/rtsp-server/rtsp-media-factory-uri.h:
3149 * gst/rtsp-server/rtsp-media-factory.h:
3150 * gst/rtsp-server/rtsp-media.h:
3151 * gst/rtsp-server/rtsp-mount-points.h:
3152 * gst/rtsp-server/rtsp-onvif-client.h:
3153 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3154 * gst/rtsp-server/rtsp-onvif-media.h:
3155 * gst/rtsp-server/rtsp-onvif-server.h:
3156 * gst/rtsp-server/rtsp-params.h:
3157 * gst/rtsp-server/rtsp-permissions.h:
3158 * gst/rtsp-server/rtsp-sdp.h:
3159 * gst/rtsp-server/rtsp-server-prelude.h:
3160 * gst/rtsp-server/rtsp-server.h:
3161 * gst/rtsp-server/rtsp-session-media.h:
3162 * gst/rtsp-server/rtsp-session-pool.h:
3163 * gst/rtsp-server/rtsp-session.h:
3164 * gst/rtsp-server/rtsp-stream-transport.h:
3165 * gst/rtsp-server/rtsp-stream.h:
3166 * gst/rtsp-server/rtsp-thread-pool.h:
3167 * gst/rtsp-server/rtsp-token.h:
3168 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3169 We need different export decorators for the different libs.
3170 For now no actual change though, just rename before the release,
3171 and add prelude headers to define the new decorator to GST_EXPORT.
3173 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3175 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3176 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3177 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3179 === release 1.13.90 ===
3181 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3187 * gst-rtsp-server.doap:
3191 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3193 * gst/rtsp-server/rtsp-media-factory.c:
3194 * gst/rtsp-server/rtsp-permissions.c:
3195 permissions: add Since tags and example for new API
3197 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3199 * docs/libs/gst-rtsp-server-sections.txt:
3200 * gst/rtsp-server/rtsp-media-factory.c:
3201 * gst/rtsp-server/rtsp-media-factory.h:
3202 * gst/rtsp-server/rtsp-permissions.c:
3203 * gst/rtsp-server/rtsp-permissions.h:
3204 * tests/check/gst/permissions.c:
3205 permissions: more bindings-friendly API
3206 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3208 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3211 meson: enable more warnings
3213 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3215 * gst/rtsp-server/rtsp-client.c:
3216 rtsp-client: Place netaddress meta on packets received via TCP
3217 This allows us to later map signals from rtpbin/rtpsource back to the
3218 corresponding stream transport, and allows to do keep-alive based on
3219 RTCP packets in case of TCP media transport.
3220 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3222 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3224 * gst/rtsp-sink/gstrtspclientsink.c:
3225 rtspclientsink: if OPEN failed, unqueue next command
3226 As READY_TO_PAUSED can no longer return async, the RECORD
3227 command will be queued before the OPEN command fails
3228 (for example in case the server could not be connected),
3229 and record then waits for ever.
3230 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3232 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3234 * gst/rtsp-sink/gstrtspclientsink.c:
3235 rtspclientsink: fix retrieval of custom payloader caps
3236 If a bin is passed as the custom payloader, the caps of
3237 its factory will be empty, the correct way to obtain the caps
3238 is to query its sinkpad.
3240 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3242 * gst/rtsp-sink/gstrtspclientsink.c:
3243 rtspclientsink: fix extra unref of custom payloader
3245 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3247 * gst/rtsp-sink/gstrtspclientsink.c:
3248 rspclientsink: fix recent code indentation
3250 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3252 * gst/rtsp-sink/gstrtspclientsink.c:
3253 rtspclientsink: add missing get_type prototype
3255 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3257 * gst/rtsp-sink/gstrtspclientsink.c:
3258 rtspclientsink: allow setting payloader as pad property
3259 This was a FIXME item, and can be quite useful, also
3260 allowing to specify payloader properties from the command
3261 line, which is always nice.
3262 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3264 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3266 * gst/rtsp-server/rtsp-media.c:
3267 rtsp-media: Replace g_print() log line
3268 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3270 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3272 * gst/rtsp-server/rtsp-media.c:
3273 * tests/check/gst/rtspclientsink.c:
3274 rtsp-media: fix RECORD getting stuck
3275 The test_record case was working because async=false had
3276 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3277 but that was incorrect, as it should not be needed.
3278 Removing async=false made the test fail as expected, this is
3279 fixed by not trying to preroll when preparing the media for
3280 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3281 and our peer will not start sending media until it has received
3282 a response to that request, and sent and received a response
3283 to RECORD as well, thus obviously preventing preroll.
3284 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3286 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3288 * gst/rtsp-server/rtsp-auth.c:
3289 rtsp-auth: fix set_tls_authentication_mode annotation
3291 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3293 * gst/rtsp-server/rtsp-onvif-media.c:
3294 rtp-server: remove redefined variable
3295 res is a boolean variable which is defined in the function scope and
3296 redefined, with no reason, in the loop scope. This patch removes the
3298 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3300 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3302 * gst/rtsp-server/rtsp-media.c:
3303 * gst/rtsp-server/rtsp-stream.c:
3304 * gst/rtsp-server/rtsp-stream.h:
3305 stream: Add functions for checking if stream is receiver or sender
3306 ...and replace all checks for RECORD in GstRTSPMedia which are really
3307 for "sender-only". This way the code becomes more generic and introducing
3308 support for onvif-backchannel later on will require no changes in
3311 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3313 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3314 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3315 onvif: Make requires_backchannel() public
3316 ...in order to let subclasses building the onvif part of the pipeline
3317 check whether backchannel shall be included or not.
3319 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3321 * gst/rtsp-server/rtsp-onvif-media.c:
3322 rtsp-server: Switch around sendonly/recvonly attributes
3323 They are wrong in the ONVIF streaming spec. The backchannel should be
3324 recvonly and the normal media should be sendonly: direction is always
3325 from the point of view of the SDP offerer (the server) according to
3328 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3330 * docs/libs/gst-rtsp-server-docs.sgml:
3331 * docs/libs/gst-rtsp-server-sections.txt:
3332 * examples/.gitignore:
3333 * examples/Makefile.am:
3334 * examples/test-onvif-backchannel.c:
3335 * gst/rtsp-server/Makefile.am:
3336 * gst/rtsp-server/rtsp-media.h:
3337 * gst/rtsp-server/rtsp-onvif-client.c:
3338 * gst/rtsp-server/rtsp-onvif-client.h:
3339 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3340 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3341 * gst/rtsp-server/rtsp-onvif-media.c:
3342 * gst/rtsp-server/rtsp-onvif-media.h:
3343 * gst/rtsp-server/rtsp-onvif-server.c:
3344 * gst/rtsp-server/rtsp-onvif-server.h:
3345 * gst/rtsp-server/rtsp-sdp.c:
3346 * gst/rtsp-server/rtsp-sdp.h:
3347 rtsp: Add support for ONVIF backchannel
3348 This adds a new RTSP server, client, media-factory and media subclass
3349 for handling the specifics of the backchannel. Ideally this later can be
3350 extended with other ONVIF specific features.
3352 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3354 * gst/rtsp-server/rtsp-media.c:
3355 rtsp-media: Add support for sending+receiving medias
3356 We need to add an appsrc/appsink in that case because otherwise the
3357 media bin will be a sink and a source for rtpbin, causing a pipeline
3359 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3361 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3367 === release 1.13.1 ===
3369 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3373 * gst-rtsp-server.doap:
3377 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3379 * gst/rtsp-server/rtsp-session-pool.c:
3380 session-pool: remove nullable return annotation
3381 create_watch can only return NULL from the API guards, no
3384 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3386 * gst/rtsp-server/rtsp-media-factory.c:
3387 * gst/rtsp-server/rtsp-media.c:
3388 set_clock functions: Add nullable annotations
3390 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3392 * gst/rtsp-server/rtsp-auth.c:
3393 * gst/rtsp-server/rtsp-client.c:
3394 * gst/rtsp-server/rtsp-media-factory.c:
3395 * gst/rtsp-server/rtsp-media.c:
3396 * gst/rtsp-server/rtsp-mount-points.c:
3397 * gst/rtsp-server/rtsp-server.c:
3398 * gst/rtsp-server/rtsp-session-media.c:
3399 * gst/rtsp-server/rtsp-session-pool.c:
3400 * gst/rtsp-server/rtsp-session.c:
3401 * gst/rtsp-server/rtsp-stream-transport.c:
3402 * gst/rtsp-server/rtsp-stream.c:
3403 * gst/rtsp-server/rtsp-thread-pool.c:
3404 All around: add annotations and API guards
3406 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3408 * tests/test-cleanup.c:
3409 test-cleanup: bind any port
3410 The meson test suite runs tests in parallel, trying to bind
3411 a single port made the test fail.
3413 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3416 meson: make version numbers ints and fix int/string comparison
3417 WARNING: Trying to compare values of different types (str, int).
3418 The result of this is undefined and will become a hard error
3419 in a future Meson release.
3421 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3423 * gst/rtsp-server/rtsp-context.c:
3424 gst_rtsp_context_get_current: add (skip) annotation
3425 The return value type is defined with G_DEFINE_POINTER_TYPE,
3426 and gi emits the following warning:
3427 Invalid non-constant return of bare structure or union; register as
3428 boxed type or (skip)
3430 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3432 * gst/rtsp-server/rtsp-client.c:
3433 rtsp-client: add type annotations
3434 gi doesn't seem to be able to figure out the type of the
3435 signal parameters when defined with G_DEFINE_POINTER_TYPE
3437 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3440 autotools: use -fno-strict-aliasing where supported
3441 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3443 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3446 meson: use -fno-strict-aliasing where supported
3447 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3449 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3451 * gst/rtsp-server/rtsp-mount-points.c:
3452 mount-points: bail out of loop again when matching mount points
3453 Previous patch led to us iterating the entire sequence. Bail out
3454 of the loop again if we have a match but are moving away from it.
3455 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3457 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3459 * tests/check/gst/mountpoints.c:
3460 tests: mountpoints: add more checks for mount point path matching
3461 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3463 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3465 * gst/rtsp-server/rtsp-mount-points.c:
3466 mount-points: fix matching of paths where there's also an entry with a common prefix
3467 e.g. with the following mount points
3471 _match() would not match /raw/video and /raw/snapshot correctly.
3472 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3474 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3476 * docs/libs/gst-rtsp-server-sections.txt:
3477 * gst/rtsp-server/rtsp-permissions.c:
3478 * gst/rtsp-server/rtsp-permissions.h:
3479 * tests/check/gst/permissions.c:
3480 permissions: add some new API to make this usable from bindings
3481 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3483 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3485 * gst/rtsp-server/rtsp-token.c:
3486 rtsp-token: annotate constructors for bindings
3487 This maps _new_empty() to _new(), which also makes RTSPToken()
3488 work properly now. Since this API wasn't usable from bindings
3489 before, this should hopefully be fine.
3490 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3492 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3494 * docs/libs/gst-rtsp-server-sections.txt:
3495 * gst/rtsp-server/rtsp-token.c:
3496 * gst/rtsp-server/rtsp-token.h:
3497 * tests/check/gst/token.c:
3498 rtsp-token: add some API to set fields from bindings
3499 The existing functions are all vararg-based and as such
3500 not usable from bindings.
3501 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3503 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3505 * tests/check/gst/rtspclientsink.c:
3506 * tests/check/gst/rtspserver.c:
3507 * tests/check/gst/sessionpool.c:
3508 * tests/check/gst/stream.c:
3509 tests: fix indentation
3512 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3514 * tests/check/gst/rtspserver.c:
3515 tests: rtspserver: fix another ref leak
3516 Even if this didn't show up in valgrind.
3518 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3520 * tests/check/gst/rtspclientsink.c:
3521 tests: rtspclientsink: fix leak
3523 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3525 * tests/check/gst/rtspserver.c:
3526 test: rtspserver: plug memory leak in test_no_session_timeout
3527 In test_no_session_timeout, unref the rtsp session object when the
3529 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3531 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3533 * gst/rtsp-sink/gstrtspclientsink.c:
3534 rtpsclientsink: Initialize and clear newly added mutex and cond
3535 While it *did* work, glib would automatically create new mutex and cond
3536 ... which never got freed
3538 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3540 * gst/rtsp-server/rtsp-stream.c:
3541 rtsp-stream: Set multicast TTL on the multicast sockets
3542 And not if we do unicast UDP.
3543 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3545 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3547 * gst/rtsp-server/rtsp-stream.c:
3548 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3549 In the multicast case (as in test-multicast, not test-multicast2), the
3550 address could be allocated/reserved (and thus set) already without
3551 allocating the actual socket. We need to allocate the socket here still
3552 instead of just claiming that it was already allocated.
3553 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3555 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3557 * gst/rtsp-sink/gstrtspclientsink.c:
3558 * gst/rtsp-sink/gstrtspclientsink.h:
3559 rtspclientsink: Use the new rtsp-stream API
3560 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3562 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3564 * gst/rtsp-sink/gstrtspclientsink.c:
3565 * gst/rtsp-sink/gstrtspclientsink.h:
3566 rtspclientsink: Wait until OPEN has been scheduled
3567 Make sure that the sink thread has started opening connection
3568 to the server before continuing.
3569 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3571 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3574 Automatic update of common submodule
3575 From e8c7a71 to 3fa2c9e
3577 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3579 * gst/rtsp-server/rtsp-media.c:
3580 * gst/rtsp-server/rtsp-session-media.c:
3581 * gst/rtsp-server/rtsp-stream.c:
3582 rtsp-server: Minor doc fixes
3585 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3588 * tests/Makefile.am:
3589 tests: disable all tests when --disable-tests is used
3590 Move conditional subdir include into top level.
3591 Based on patch by: Joel Holdsworth
3592 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3594 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3597 * meson_options.txt:
3598 * tests/meson.build:
3599 meson: build more tests and add options to disable tests and examples
3601 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3603 * gst/rtsp-server/rtsp-session.c:
3604 Fix build when -Werror=deprecated-declarations is on
3605 As gst_rtsp_session_next_timeout is deprecated.
3607 ../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]
3608 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3610 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3611 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3612 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3615 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3618 Automatic update of common submodule
3619 From 3f4aa96 to e8c7a71
3621 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3623 * tests/check/gst/media.c:
3624 check/media: Add seekability test case: not all streams are active
3625 Media contains two streams but only one is complete and prepared
3627 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3629 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3631 * gst/rtsp-server/rtsp-stream.c:
3632 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3633 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3635 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3637 * gst/rtsp-server/rtsp-media.c:
3638 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
3639 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3641 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
3644 meson: remove vs_module_defs_dir variable which is no longer needed
3646 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
3648 * gst/rtsp-server/rtsp-session.h:
3651 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
3654 * gst/rtsp-server/meson.build:
3656 * win32/common/libgstrtspserver.def:
3657 win32: remove .def file with exports
3658 They're no longer needed, symbol exporting is now explicit
3659 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
3661 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3664 autotools: stop controlling symbol visibility with -export-symbols-regex
3665 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
3666 This should result in consistent behaviour for the autotools and
3669 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
3671 * gst/rtsp-server/rtsp-media.h:
3672 * gst/rtsp-server/rtsp-server.h:
3673 * gst/rtsp-server/rtsp-session.c:
3674 * gst/rtsp-server/rtsp-session.h:
3675 rtsp-server: add missing GST_EXPORT and export deprecated funcs
3677 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
3679 * tests/check/gst/media.c:
3680 check: Add seekability testing on medias
3681 Make sure that once GstRTSPMedia are prepared they returned
3682 the expected seekability results
3683 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3685 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
3687 * docs/libs/gst-rtsp-server-sections.txt:
3688 * gst/rtsp-server/rtsp-media.c:
3689 * gst/rtsp-server/rtsp-stream.c:
3690 * gst/rtsp-server/rtsp-stream.h:
3691 * win32/common/libgstrtspserver.def:
3692 rtsp-media: Enable seeking query before pipeline is complete
3693 SDP are now provided *before* the pipeline is fully complete. In order
3694 to know whether a media is seekable or not therefore requires asking
3695 the invididual streams.
3696 API: gst_rtsp_stream_seekable
3697 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3699 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
3701 * gst/rtsp-server/rtsp-media.c:
3702 rtsp-media: Fix handling in default_unsuspend()
3703 Handle the case when streams are not blocked and media
3704 is suspended from PAUSED.
3705 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
3706 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3708 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
3710 * tests/check/gst/media.c:
3711 check/media: Fix thread pool leak.
3712 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
3713 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3715 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
3717 * gst/rtsp-server/rtsp-media.c:
3718 rtsp-media: Removed fakesink elements
3719 There is not need of adding fakesink elements to the media
3720 pipeline in the dynamic-payloader case.
3721 The media pipeline itself is dynamically updated with
3722 the receiver and sender parts that are based on the client
3723 transport information known after SETUP has been received.
3724 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
3725 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3727 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
3729 * gst/rtsp-server/rtsp-media.c:
3730 rtsp-media: Corrected ASYNC_DONE handling
3731 Media is complete when all the transport based parts are
3732 added to the media pipeline. At this point ASYNC_DONE is
3733 posted by the media pipeline and media is ready to enter
3735 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
3736 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3738 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
3740 * tests/check/gst/media.c:
3741 check/media: Check that prepared media can provide a SDP
3742 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
3744 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
3746 * gst/rtsp-server/rtsp-client.c:
3747 rtsp-client: Don't leak addr
3750 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
3752 * gst/rtsp-server/rtsp-client.c:
3753 * gst/rtsp-server/rtsp-session-media.c:
3754 * gst/rtsp-server/rtsp-stream.c:
3757 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
3759 * gst/rtsp-server/rtsp-media.c:
3760 rtsp-media: Don't unblock with remaining dynamic payloaders
3761 If we still have some dynamic paylaoders which haven't posted
3762 no-more-pads yet, don't go to PREPARED if one of the streams
3764 The risk was that we would end up not exposing/using all specified
3766 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
3767 then it will take a bit more time to start. But only if those 3
3768 conditions are present.
3769 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3771 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
3773 * gst/rtsp-server/rtsp-media.c:
3776 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
3778 * gst/rtsp-server/rtsp-media.c:
3779 rtsp-media: Don't set float on a gint64 variable
3780 Just use 0. Fixes 'undefined' behaviour from clang
3782 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
3784 * gst/rtsp-server/rtsp-media.c:
3785 rtsp-media: Fix previous commit
3786 We only want to count dynamic payloaders
3788 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
3790 * gst/rtsp-server/rtsp-media.c:
3791 * tests/check/gst/media.c:
3792 rtsp-media: Handle multiple dynamic elements
3793 If we have more than one dynamic payloader in the pipeline, we need
3794 to wait until the *last* one emits 'no-more-pads' before switching
3796 Failure to do so would result in a race where some of the streams
3797 wouldn't properly be prepared
3798 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3800 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
3802 * win32/common/libgstrtspserver.def:
3803 win32: Fix exported symbols list
3805 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3807 * gst/rtsp-server/rtsp-stream.c:
3808 rtsp-stream: Only update the RTP udpsink if it actually exists
3809 For send-only streams it does not exist, but the RTCP udpsink might.
3811 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
3813 * win32/common/libgstrtspserver.def:
3814 win32: Update exports
3816 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
3818 * gst/rtsp-server/rtsp-media.c:
3819 * gst/rtsp-server/rtsp-stream.c:
3820 * gst/rtsp-server/rtsp-stream.h:
3821 rtsp-media: seek on media pipelines that are complete
3822 Make sure that a seek is performed on pipelines that
3823 contain at least one sink element.
3824 Change-Id: Icf398e10add3191d104b1289de612412da326819
3825 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3827 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
3829 * gst/rtsp-server/rtsp-client.c:
3830 * gst/rtsp-server/rtsp-media.c:
3831 * gst/rtsp-server/rtsp-media.h:
3832 * gst/rtsp-server/rtsp-stream.c:
3833 * gst/rtsp-server/rtsp-stream.h:
3834 * tests/check/gst/client.c:
3835 * tests/check/gst/media.c:
3836 * tests/check/gst/rtspserver.c:
3837 * tests/check/gst/stream.c:
3838 Dynamically reconfigure pipeline in PLAY based on transports
3839 The initial pipeline does not contain specific transport
3840 elements. The receiver and the sender parts are added
3842 If the media is shared, the streams are dynamically
3843 reconfigured after each PLAY.
3844 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3846 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
3848 * gst/rtsp-server/rtsp-stream.c:
3849 rtsp-stream: obtain stream position from pad
3850 If no sinks have been added yet, obtain the current and
3851 the stop position of the stream from the send_src pad.
3852 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
3853 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3855 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
3857 * gst/rtsp-server/rtsp-session-media.c:
3858 * gst/rtsp-server/rtsp-session-media.h:
3859 rtsp-session-media: add function to get a list of transports
3860 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
3861 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3863 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
3865 * gst/rtsp-server/rtsp-stream.c:
3866 * gst/rtsp-server/rtsp-stream.h:
3867 rtsp-stream: add functions to get rtp and rtcp multicast sockets
3868 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
3869 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3871 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
3873 * gst/rtsp-server/rtsp-stream.c:
3874 stream: set async=sync=false only for RTCP appsink
3875 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
3876 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3878 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
3880 * gst/rtsp-server/rtsp-media.c:
3881 rtsp-media: return minimum value in query position case
3882 The minimum position should be returned as we are interested
3883 in the whole interval.
3884 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
3885 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3887 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
3889 * gst/rtsp-server/rtsp-session.c:
3890 * tests/check/gst/rtspserver.c:
3891 rtsp-session: Handle the case when timeout=0
3892 According to the documentation, a timeout of value 0 means
3893 that the session never timeouts. This adds handling of that.
3894 If timeout=0 we just return with a -1 from
3895 gst_rtsp_session_next_timeout_usec ().
3896 https://bugzilla.gnome.org/show_bug.cgi?id=785058
3898 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3900 * gst/rtsp-sink/gstrtspclientsink.c:
3901 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
3902 https://bugzilla.gnome.org/show_bug.cgi?id=785024
3904 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3906 * docs/libs/gst-rtsp-server-sections.txt:
3907 * gst/rtsp-server/rtsp-media-factory.c:
3908 docs: add media factory transport mode accessors
3909 and fix the documentation for the return value of the getter
3911 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
3913 * gst/rtsp-server/rtsp-client.c:
3914 rtsp-client: unref 'pipelined_requests' in finalize
3915 The hash table priv->pipelined_requests is not unref:ed in the
3916 finalize funktion. Make sure it is.
3917 https://bugzilla.gnome.org/show_bug.cgi?id=788704
3919 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
3921 * gst/rtsp-server/rtsp-media.c:
3922 rtsp-media: Initialize scalar variable
3925 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
3927 * win32/common/libgstrtspserver.def:
3928 win32: Update export file
3930 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
3932 * gst/rtsp-server/rtsp-client.c:
3933 * gst/rtsp-server/rtsp-media.c:
3934 * gst/rtsp-server/rtsp-media.h:
3935 Start support for RTSP 2.0
3936 This adds basic support for new 2.0 features, though the protocol is
3937 subposdely backward incompatible, most semantics are the sames.
3940 * version negotiation
3941 * pipelined requests support
3942 * Media-Properties support
3943 * Accept-Ranges support
3945 * gst_rtsp_media_seekable
3946 The RTSP methods that have been removed when using 2.0 now return
3948 https://bugzilla.gnome.org/show_bug.cgi?id=781446
3950 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3952 * gst/rtsp-server/rtsp-stream.c:
3953 stream: Use stream duration as stream-stop if segment was not configured with a stop
3954 Allowing client to know stream duration when no seeking happened.
3955 https://bugzilla.gnome.org/show_bug.cgi?id=783435
3957 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
3959 * gst/rtsp-server/rtsp-media-factory.c:
3960 rtsp-media-factory: Don't cache any media if NULL was returned as key
3961 The docs already mentioned this, but we actually stored it in the hash
3962 table with key==NULL and leaked its reference forever.
3964 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
3966 * gst/rtsp-sink/gstrtspclientsink.c:
3967 * gst/rtsp-sink/gstrtspclientsink.h:
3968 rtspclientsink: Use a mutex for protecting against concurrent send/receives
3969 This is a simple port of:
3970 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
3971 * c438545dc9e2f14f657bc0ef261fff726449867b
3972 * cd17c71dcea5c9310d21f1347c7520983e5869ac
3973 in gst-plugins-good.
3975 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
3977 * gst/rtsp-server/rtsp-sdp.c:
3978 sdp: fix Memory leak in error case
3979 https://bugzilla.gnome.org/show_bug.cgi?id=787059
3981 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3983 * pkgconfig/meson.build:
3984 meson: don't install -uninstalled.pc file
3985 https://bugzilla.gnome.org/show_bug.cgi?id=786457
3987 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
3990 Automatic update of common submodule
3991 From 48a5d85 to 3f4aa96
3993 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3995 * gst/rtsp-server/rtsp-client.c:
3996 rtsp-client: Fix typo in debug message
3998 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4001 meson: hide symbols by default unless explicitly exported
4003 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4005 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4006 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4007 Fixes meson warning about undefined @srcdir@.
4009 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4011 * tests/meson.build:
4012 meson: skip tests on windows for now
4013 As we do in the other modules. As libgstcheck is currently not
4014 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4015 the subproject 'gstreamer' does not exist"" Meson error.
4017 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4019 * gst/rtsp-server/rtsp-stream.c:
4020 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4021 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4022 multiudpsink's last-sample always comes from the payloader. Which
4023 is wrong if auxiliary streams are multiplexed in the same stream.
4024 So check the buffer's ssrc against the caps'ssrc before to use its
4025 seqnum. If not the same ssrc just use the payloader as done prior
4026 the commit above or when there is no last-sample yet.
4027 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4029 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4032 meson: Allow using glib as a subproject
4034 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4037 meson: fix with-package-name option
4038 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4040 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4043 Distribute meson_options.txt
4045 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4048 And config.h.meson is no longer dist either
4050 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4054 meson: config.h.meson is no longer needed
4056 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4058 * tests/check/meson.build:
4059 * tests/meson.build:
4060 meson: Fix building tests and activate them again
4062 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4064 * tests/check/meson.build:
4065 meson: Do not use path separator in test names
4066 Avoiding warnings like:
4067 WARNING: Target "elements/audioamplify" has a path separator in its name.
4069 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4072 * meson_options.txt:
4073 meson: add options to set package name and origin
4074 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4076 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4078 * gst/rtsp-server/rtsp-address-pool.h:
4079 * gst/rtsp-server/rtsp-auth.h:
4080 * gst/rtsp-server/rtsp-client.h:
4081 * gst/rtsp-server/rtsp-context.h:
4082 * gst/rtsp-server/rtsp-media-factory-uri.h:
4083 * gst/rtsp-server/rtsp-media-factory.h:
4084 * gst/rtsp-server/rtsp-media.h:
4085 * gst/rtsp-server/rtsp-mount-points.h:
4086 * gst/rtsp-server/rtsp-params.h:
4087 * gst/rtsp-server/rtsp-permissions.h:
4088 * gst/rtsp-server/rtsp-sdp.h:
4089 * gst/rtsp-server/rtsp-server.h:
4090 * gst/rtsp-server/rtsp-session-media.h:
4091 * gst/rtsp-server/rtsp-session-pool.h:
4092 * gst/rtsp-server/rtsp-session.h:
4093 * gst/rtsp-server/rtsp-stream-transport.h:
4094 * gst/rtsp-server/rtsp-stream.h:
4095 * gst/rtsp-server/rtsp-thread-pool.h:
4096 * gst/rtsp-server/rtsp-token.h:
4097 Mark symbols explicitly for export with GST_EXPORT
4099 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4102 * gst/rtsp-sink/Makefile.am:
4103 Remove plugin specific static build option
4104 Static and dynamic plugins now have the same interface. The standard
4105 --enable-static/--enable-shared toggle are sufficient.
4107 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4113 === release 1.12.0 ===
4115 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4121 * gst-rtsp-server.doap:
4125 === release 1.11.91 ===
4127 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4133 * gst-rtsp-server.doap:
4137 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4140 Automatic update of common submodule
4141 From 60aeef6 to 48a5d85
4143 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4145 * gst/rtsp-server/rtsp-media-factory.c:
4146 * gst/rtsp-server/rtsp-media.c:
4147 * gst/rtsp-server/rtsp-session.c:
4148 * gst/rtsp-server/rtsp-stream.c:
4149 gi: Fix some annotations and docstrings
4151 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4153 * gst/rtsp-server/meson.build:
4155 * meson_options.txt:
4158 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4162 Automatic update of common submodule
4163 From 39ac2f5 to 60aeef6
4165 === release 1.11.90 ===
4167 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4173 * gst-rtsp-server.doap:
4177 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4179 * examples/test-launch.c:
4180 examples: make test-launch pipeline shared by default as well
4182 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4184 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4185 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4186 Just the build dir is not going to work for srcdir!=builddir.
4188 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4191 meson: Update version
4193 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4198 === release 1.11.2 ===
4200 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4206 * gst-rtsp-server.doap:
4209 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4212 meson: dist meson build files
4213 Ship meson build files in tarballs, so people who use tarballs
4214 in their builds can start playing with meson already.
4216 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4218 * examples/test-record.c:
4219 examples/test-record: Add extra line to initial printout
4220 Add an example line of how to deliver a stream to the
4223 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4225 * gst/rtsp-server/rtsp-client.c:
4226 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4227 If there is no Content-Length header, no body would be allocated and the
4228 '\0' would also not be appended to the body.
4230 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4232 * gst/rtsp-server/rtsp-client.c:
4233 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4234 While they logically have 0 bytes length, GstRTSPConnection is appending
4235 a '\0' to everything making the size be 1 instead.
4237 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4242 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4244 * gst/rtsp-server/rtsp-session.c:
4245 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4246 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4249 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4254 === release 1.11.1 ===
4256 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4262 * gst-rtsp-server.doap:
4263 * win32/common/libgstrtspserver.def:
4266 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4268 * gst/rtsp-server/rtsp-stream.c:
4269 rtsp-stream: corrected if-statement in _get_server_port()
4270 This bug was accidentally introduced while fixing a segfault
4271 in _get_server_port() function.
4272 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4274 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4276 * gst/rtsp-server/rtsp-stream.c:
4277 * tests/check/gst/stream.c:
4278 rtsp-stream: fixed segmenation fault in _get_server_port()
4279 Calling function gst_rtsp_stream_get_server_port() results in
4280 segmenation fault in the RTP/RTSP/TCP case.
4281 Port that the server will use to receive RTCP makes only
4282 sense in the UDP case, however the function should handle
4283 the TCP case in a nicer way.
4284 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4286 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4288 * gst/rtsp-server/rtsp-media-factory.c:
4289 dosc: Fix a little typo
4290 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4292 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4294 * pkgconfig/Makefile.am:
4295 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4296 * pkgconfig/meson.build:
4297 meson: generate pkg-config -uninstalled pc files
4298 Generating those files is useful for users building the GStreamer stack
4299 using meson and having to link it to another project which is still
4300 using the autotools.
4301 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4303 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4305 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4306 pkgconfig: fix -uninstalled pc file
4307 pcfiledir was never defined so the paths were wrong.
4308 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4310 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4312 * gst/rtsp-server/rtsp-stream.c:
4313 * tests/check/gst/rtspserver.c:
4314 rtsp-stream: Fixed TCP transport case
4315 Make sure that the appsink element is actually added to
4316 the bin before trying to link it with the elements in it.
4317 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4319 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4325 Remove generated .spec file
4326 Likely extremely bitrotten, and we should not ship this anyway.
4328 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4331 Automatic update of common submodule
4332 From f980fd9 to 39ac2f5
4334 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4336 * gst/rtsp-server/rtsp-media.c:
4337 media: Fix pt map caps
4338 Since decryption is handled within rtpbin, all outcoming stream
4339 caps will be application/x-rtp (i.e. regular rtp)
4340 Fixes RECORD with SRTP streams
4342 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4344 * gst/rtsp-server/rtsp-media-factory.c:
4345 media-factory: Create media objects with the proper transport mode
4346 The function called immediately afterwards (collect_streams()) will
4347 need it to work properly
4349 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4351 * gst/rtsp-server/rtsp-auth.c:
4352 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4354 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4356 * gst/rtsp-server/rtsp-media-factory.c:
4357 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4358 We're going to put a pipeline into a pipeline otherwise, which is not
4361 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4363 * gst/rtsp-server/rtsp-media.c:
4364 media: Fix race condition around finish_unprepare() if called multiple time
4365 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4367 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4369 * gst/rtsp-sink/gstrtspclientsink.c:
4370 rtspclientsink: Don't leave stale pointer after unref
4371 Fix a warning on shutdown - don't keep a pointer to an
4372 alread-unreffed object.
4374 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4377 common: use https protocol for common submodule
4378 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4380 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4382 * gst/rtsp-server/rtsp-stream.c:
4383 stream: block the output of rtpbin instead of the source pipeline
4384 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4385 detection of the srtp rollover counter to add to the SDP.
4386 Unfortunately, it was incomplete for live pipelines where the logic
4387 blocks the source bin before creating the SDP and thus would never have
4388 the necessary informaiton to create a correct SDP with srtp encryption.
4389 Move the pad blocks to rtpbin's output pads instead so that the
4390 necessary information can be created before we need the information for
4392 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4394 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4396 * gst/rtsp-server/rtsp-client.c:
4397 rtsp-client: add IDLE timeout, before session exists
4398 The RTSP server will not timeout an idle RTSP connection
4399 (note this is different from doing timeout on a RTSP
4401 At least for Apache this is a problem when running RTSP over
4402 HTTPS since it uses one of the threads (there is a rather
4403 limited number) that are available for handling requests.
4404 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4406 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4411 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4413 * gst/rtsp-server/rtsp-stream.c:
4414 rtsp-stream: Set close-socket FALSE on UDP src:es
4415 With this RTSP server can use the sockets independent on the udpsrc
4417 When the udp src is finalized it will unref socket and when g_socket
4418 is finalized the socket will be closed.
4419 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4421 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4423 * gst/rtsp-sink/gstrtspclientsink.c:
4424 rtspclientsink: Move to new helper function to parse authentication responses
4425 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4427 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4429 * examples/Makefile.am:
4430 * examples/test-auth-digest.c:
4431 * gst/rtsp-server/rtsp-auth.c:
4432 * gst/rtsp-server/rtsp-auth.h:
4433 * win32/common/libgstrtspserver.def:
4434 rtsp-auth: Add support for Digest authentication
4435 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4437 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4440 * gst/rtsp-server/meson.build:
4442 * tests/check/meson.build:
4444 * win32/common/libgstrtspserver.def:
4445 Enable building with MSVC
4446 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4448 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4451 meson: gstreamer gst_check_dep does not exist on windows
4453 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4455 * gst/rtsp-server/rtsp-client.c:
4456 client: update do_send_message to match type GstRTSPClientSendFunc
4457 This type mismatch fails building with MSVC
4458 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4460 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4462 * gst/rtsp-server/rtsp-sdp.c:
4463 rtsp-sdp: Fix indentation
4465 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4467 * gst/rtsp-server/rtsp-media.c:
4468 rtsp-media: Only signal "new-state" if the state has actually changed
4469 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4471 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4473 * gst/rtsp-server/rtsp-client.c:
4474 * gst/rtsp-server/rtsp-client.h:
4475 client: emit signal in the beginning of each rtsp request
4476 These signals let the application validate the requests, configure the
4477 media/stream in a certain way and also generate error status code in
4478 case of error or bad request.
4479 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4481 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4484 meson: update version
4486 === release 1.11.0 ===
4488 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4493 === release 1.10.0 ===
4495 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4501 * gst-rtsp-server.doap:
4504 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4506 * tests/check/gst/rtspserver.c:
4507 * tests/check/gst/stream.c:
4508 tests: try to avoid using the same ports in different tests
4509 Causes problems with client multicast tests otherwise if
4510 tests are run in parallel.
4511 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4513 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4515 * tests/check/gst/client.c:
4516 tests: client: use fail_unless_equals_foo() for better failure reporting
4518 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4520 * gst/rtsp-server/rtsp-client.c:
4521 rtsp-client: Session filter in unwatch session
4522 Call session filter with filter_session_media as paramer in
4523 client_unwatch_session if using drop_backlog = FALSE.
4524 In client_unwatch_session its allowed to grow the watchs backlog.
4525 If using drop_backlog = FALSE and the backlog is full it will cause
4526 a deadlock when setting session media state to NULL
4527 if the backlog is not allowed to grow.
4528 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4530 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4533 meson: add fallbacks for gst modules
4536 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4538 * gst/rtsp-server/rtsp-client.c:
4539 rtsp-client: Fix factory leaking in find_media() in error cases
4540 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4542 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4544 * gst/rtsp-server/rtsp-stream.c:
4545 stream: Fix randomly missing streams from SDP with dynamic elements
4546 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4547 "pad-added" signal. In that case priv->srcpad could already have its caps,
4548 and they'll be sent to priv->send_src[0] pad. That means that when it
4549 connects "notify::caps" signal, that pad could already have received its
4550 caps and the signal won't be emitted anymore.
4551 In that case priv->caps stay to NULL and when building the SDP that stream
4552 gets ignored. Leading to missing video or audio when playing in client side.
4553 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4555 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4558 meson: update version
4560 === release 1.9.90 ===
4562 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4568 * gst-rtsp-server.doap:
4571 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4573 * gst/rtsp-server/rtsp-media-factory.c:
4574 * gst/rtsp-server/rtsp-media.c:
4575 * gst/rtsp-server/rtsp-stream.c:
4576 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4577 To prevent any possibly confusion with IPs or anything else.
4578 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4580 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4582 * gst/rtsp-server/rtsp-media-factory.c:
4583 * gst/rtsp-server/rtsp-media.c:
4584 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4585 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4587 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4590 configure: Depend on gstreamer 1.9.2.1
4592 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4596 Automatic update of common submodule
4597 From b18d820 to f980fd9
4599 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4603 Automatic update of common submodule
4604 From 6f2d209 to b18d820
4606 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4608 * gst/rtsp-server/rtsp-stream.c:
4609 rtsp-stream: Remove unused _locked() variant of a function
4610 It was added during refactoring.
4612 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4614 * gst/rtsp-server/rtsp-stream.c:
4615 stream: cosmetic cleanup
4616 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4618 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4620 * gst/rtsp-server/rtsp-stream.c:
4621 stream: Compare IP addresses case insensitive in more places
4622 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4624 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4627 * gst/rtsp-server/rtsp-stream.c:
4628 stream: Fix leaked joined_bin
4629 There is no need to keep a strong ref on it, and _leave_bin() was
4630 setting it to NULL before calling g_clear_object() so it was leaked.
4631 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4633 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4635 * gst/rtsp-server/rtsp-stream.c:
4636 rtsp-stream: Compare IP address strings case insensitive
4637 Otherwise IPv6 addresses might fail this comparision.
4639 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
4641 * gst/rtsp-server/rtsp-stream.c:
4642 rtsp-stream: Bind multicast sockets to ANY as before
4643 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
4645 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
4647 * gst/rtsp-server/rtsp-session.c:
4648 rtsp-session: Fix segfault when doing keep-alive after removing the session
4649 If keep-alive happens after removing the session but before finalizing the
4650 stream transport, we would segfault.
4651 https://bugzilla.gnome.org/show_bug.cgi?id=750544
4653 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
4655 * gst/rtsp-server/rtsp-stream.c:
4656 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
4657 Adding them later will cause deadlocks due to
4658 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
4659 2) adding the multicast sink
4660 3) waiting for it to get data to preroll again
4661 3) never happens because the queues after the tee are full.
4663 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
4665 * gst/rtsp-server/rtsp-stream.c:
4666 rtsp-stream: Fix up various multicast related issues
4668 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
4670 * tests/check/gst/stream.c:
4671 tests: Fix compilation
4673 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4675 * gst/rtsp-server/rtsp-client.c:
4676 * gst/rtsp-server/rtsp-stream.c:
4677 * tests/check/gst/stream.c:
4678 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
4679 This is basically reverting changes introduced in commit f62a9a7,
4680 because it was introducing various regressions:
4681 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
4682 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
4683 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
4684 - If a mcast client connects, it creates a new socket in SETUP to try to respect
4685 the destination/port given by the client in the transport, and overrides the
4686 socket already set on the udpsink element. That means that if we already had a
4687 client connected, the source address on the udp packets it receives suddenly
4689 - If a 2nd mcast client connects, the destination/port in its transport is
4690 ignored but its transport wasn't updated.
4691 What this patch does:
4692 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
4693 - Always have a tee+queue when udp is enabled. This could be optimized
4694 again in a later patch, but is more complicated. If no unicast clients
4695 connects then those elements are useless, this could be also optimized
4697 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
4698 seperated from those for unicast clients. Since we already support only
4699 one mcast address, we also create only one set of elements.
4700 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4702 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4704 * gst/rtsp-server/rtsp-stream.c:
4705 stream: factor our plug_src function
4706 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4708 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4710 * gst/rtsp-server/rtsp-stream.c:
4711 stream: factor out plug_sink function
4712 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4714 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4716 * gst/rtsp-server/rtsp-stream.c:
4717 stream: small documentation clarification
4718 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4720 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4722 * gst/rtsp-server/rtsp-stream.c:
4723 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
4724 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4726 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4728 * gst/rtsp-server/rtsp-stream.c:
4729 stream: Keep a ref on joined bin
4730 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4732 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4734 * gst/rtsp-server/rtsp-stream.c:
4735 stream: code cleanup
4736 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4738 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4740 * gst/rtsp-server/rtsp-stream.c:
4741 stream: small fix in error code path
4742 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4744 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4746 * gst/rtsp-server/rtsp-stream.c:
4747 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
4748 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
4749 but keeps unit tests.
4750 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4752 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
4757 === release 1.9.2 ===
4759 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
4765 * gst-rtsp-server.doap:
4768 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
4771 * examples/meson.build:
4773 * gst/rtsp-server/meson.build:
4774 * gst/rtsp-sink/meson.build:
4776 * pkgconfig/meson.build:
4777 * tests/check/meson.build:
4778 * tests/meson.build:
4779 Add support for Meson as alternative/parallel build system
4780 https://github.com/mesonbuild/meson
4782 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
4785 * tests/check/Makefile.am:
4786 build: silence error about pthread for 'make check' in osx
4787 Fixes "clang: error: argument unused during compilation: '-pthread'"
4789 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
4791 * gst/rtsp-server/rtsp-client.c:
4792 rtsp-client: Fix leaking of media in error cases
4793 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
4794 and myself to make the media refcounting a bit easier to follow.
4795 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4797 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4799 * gst/rtsp-server/rtsp-client.c:
4800 rtsp-client: Fix leaking of session in error cases
4801 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4803 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
4806 Automatic update of common submodule
4807 From f363b32 to f49c55e
4809 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
4814 === release 1.9.1 ===
4816 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4822 * gst-rtsp-server.doap:
4825 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4828 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
4829 https://bugzilla.gnome.org/show_bug.cgi?id=767463
4831 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4834 Automatic update of common submodule
4835 From ac2f647 to f363b32
4837 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4839 * gst/rtsp-server/rtsp-sdp.c:
4840 * gst/rtsp-server/rtsp-sdp.h:
4841 * gst/rtsp-server/rtsp-stream.c:
4842 * gst/rtsp-server/rtsp-stream.h:
4843 sdp: add rollover counters for all sender SSRC
4844 We add different crypto sessions in MIKEY, one for each sender
4845 SSRC. Currently, all of them will have the same security policy, 0.
4846 The rollover counters are obtained from the srtpenc element using the
4848 https://bugzilla.gnome.org/show_bug.cgi?id=730539
4850 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
4852 * gst/rtsp-server/rtsp-media-factory.h:
4853 * gst/rtsp-server/rtsp-server.h:
4854 docs: fix some typos
4856 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
4858 * gst/rtsp-server/Makefile.am:
4859 g-i: pass compiler env to g-ir-scanner
4860 It's what introspection.mak does as well. Should
4861 fix spurious build failures on gnome-continuous
4862 (caused by g-ir-scanner getting compiler details
4863 via python which is broken in some environments
4864 so passing the compiler details bypasses that).
4866 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
4868 * gst/rtsp-server/rtsp-session.c:
4869 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
4870 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
4871 https://bugzilla.gnome.org/show_bug.cgi?id=766619
4873 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
4875 * gst/rtsp-sink/gstrtspclientsink.c:
4876 rtspclientsink: Check return value of sscanf
4877 And just make sure we always have 0/0 if we have an error
4880 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
4882 * gst/rtsp-server/rtsp-stream.c:
4883 * tests/check/gst/rtspserver.c:
4884 * tests/check/gst/stream.c:
4885 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
4886 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
4887 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
4888 - Create unit test for shared media.
4889 https://bugzilla.gnome.org/show_bug.cgi?id=764744
4891 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4893 * gst/rtsp-server/rtsp-stream.c:
4894 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
4895 For IPv6 addresses, binding to a multicast group does not work on Linux
4896 either. Always bind to ANY and then later join the multicast group.
4897 https://bugzilla.gnome.org/show_bug.cgi?id=764679
4899 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
4902 Automatic update of common submodule
4903 From 6f2d209 to ac2f647
4905 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
4907 * gst/rtsp-server/rtsp-thread-pool.c:
4908 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
4909 Clarified why it is necessary to add source information to
4910 GstRTSPThreadImpl. See the reported bug in GLib:
4911 https://bugzilla.gnome.org/show_bug.cgi?id=720186
4912 for more information.
4913 https://bugzilla.gnome.org/show_bug.cgi?id=761702
4915 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
4917 * examples/Makefile.am:
4918 examples: Clean up CFLAGS/LDADD even more
4919 The internal .la should come first and is part of LDADD, as is
4922 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
4924 * examples/Makefile.am:
4925 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
4927 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
4929 * gst/rtsp-server/Makefile.am:
4930 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
4932 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
4934 * gst/rtsp-server/rtsp-client.c:
4935 * gst/rtsp-server/rtsp-media-factory.c:
4936 * gst/rtsp-server/rtsp-media-factory.h:
4937 * gst/rtsp-server/rtsp-media.c:
4938 * gst/rtsp-server/rtsp-media.h:
4939 * gst/rtsp-server/rtsp-sdp.c:
4940 * gst/rtsp-server/rtsp-stream.c:
4941 * gst/rtsp-server/rtsp-stream.h:
4942 rtsp-server: Implement clock signalling according to RFC7273
4943 For NTP and PTP clocks we signal the actual clock that is used and signal
4944 the direct media clock offset.
4945 For all other clocks we at least signal that it's the local sender clock.
4946 This allows receivers to know which clock was used to generate the media and
4947 its RTP timestamps. Receivers can then implement network synchronization,
4948 either absolute or at least relative by getting the sender clock rate directly
4949 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
4951 https://bugzilla.gnome.org/show_bug.cgi?id=760005
4953 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
4955 * gst/rtsp-sink/gstrtspclientsink.c:
4956 rtspclientsink: Add support for setting the multicast interface
4957 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4959 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4961 * gst/rtsp-server/rtsp-media-factory.c:
4962 * gst/rtsp-server/rtsp-media-factory.h:
4963 * gst/rtsp-server/rtsp-media.c:
4964 * gst/rtsp-server/rtsp-media.h:
4965 * gst/rtsp-server/rtsp-stream.c:
4966 * gst/rtsp-server/rtsp-stream.h:
4967 rtsp-media: Add support for setting the multicast interface
4968 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4970 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
4972 * gst/rtsp-sink/gstrtspclientsink.c:
4973 rtspclientsink: use new gst_element_class_add_static_pad_template()
4974 https://bugzilla.gnome.org/show_bug.cgi?id=763196
4976 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4981 === release 1.8.0 ===
4983 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
4989 * gst-rtsp-server.doap:
4992 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
4994 * gst/rtsp-server/rtsp-stream.c:
4995 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
4996 This would get us NO_PREROLL in the bin again and break seeking.
4997 Thanks to Carlos Rafael Giani for helping to debug this!
4998 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5000 === release 1.7.91 ===
5002 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5008 * gst-rtsp-server.doap:
5011 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5013 * gst/rtsp-server/rtsp-stream.c:
5014 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5015 Without this, RECORD pipelines are broken because
5016 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5017 added later. Previously it was there earlier and due to NO_PREROLL caused the
5018 pipeline to preroll immediately
5019 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5020 as the corresponding code previously was only for PLAY pipelines.
5021 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5023 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5025 * gst/rtsp-server/rtsp-stream.c:
5026 rtsp-stream: Fix typo in the docstring
5027 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5029 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5031 * gst/rtsp-server/rtsp-stream.c:
5032 rtsp-stream: Disable multicast loopback for all our sockets
5033 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5034 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5035 loopback setting on the socket... while udpsink does which unfortunately has
5036 no effect here on Windows but on Linux.
5037 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5039 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5041 * tests/check/gst/stream.c:
5042 stream tests: added new tests
5043 Test a case when the address pool only contains multicast addresses
5044 and the client is requesting unicast udp.
5045 Added tests for multicast ports allocation.
5046 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5048 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5050 * gst/rtsp-server/rtsp-stream.c:
5051 rtsp-stream: Only bind multicast sockets to ANY on Windows
5052 On Linux it is still needed to bind to the multicast address
5053 to filter out random other packets, while on Windows binding
5054 to multicast addresses just fails.
5056 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5058 * gst/rtsp-server/rtsp-stream.c:
5059 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5060 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5061 addresses, which is something that used to work before. For unicast addresses
5062 if the pool contains none, we just allocate them as if there is no pool at
5064 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5066 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5068 * gst/rtsp-server/rtsp-client.c:
5069 * gst/rtsp-server/rtsp-stream.c:
5070 rtsp-server: Fix indentation
5072 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5074 * gst/rtsp-server/rtsp-stream.c:
5075 rtsp-stream: Don't bind the sockets to multicast addresses
5076 This works on Linux but fails completely on Windows. You're supposed
5077 to bind to ANY and then join the multicast group.
5078 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5080 === release 1.7.90 ===
5082 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5088 * gst-rtsp-server.doap:
5091 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5094 Automatic update of common submodule
5095 From b64f03f to 6f2d209
5097 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5099 * gst/rtsp-sink/gstrtspclientsink.c:
5100 * tests/check/gst/rtspclientsink.c:
5101 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5102 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5104 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5106 * tests/check/gst/media.c:
5107 * tests/check/gst/rtspclientsink.c:
5108 * tests/check/gst/rtspserver.c:
5109 * tests/check/gst/stream.c:
5110 tests: unit test fixes
5111 Removed port allocation test from the media suite.
5112 The port allocation failure is now in the stream suite.
5114 Make sure that the media is suspended after the DESCRIBE request
5115 before reconfiguring the UDP sinks.
5117 In the RECORD case we have to set async property to false
5118 for the appsink element in the test in order to make sure
5119 that the media pipeline doesn't hang in start_preroll().
5120 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5122 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5124 * gst/rtsp-server/rtsp-client.c:
5125 * gst/rtsp-server/rtsp-stream.c:
5126 * gst/rtsp-server/rtsp-stream.h:
5127 rtsp-stream: postpone UDP socket allocation until SETUP
5128 Postpone the allocation of the UDP sockets until we know
5129 what transport has been chosen by the client.
5130 Both unicast and multicast UDP sources are created in one
5132 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5134 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5136 * gst/rtsp-server/rtsp-stream.c:
5137 rtsp-stream: postpone the creation of the UDP sources
5138 Code refactoring: allocate the UDP ports after the sender and
5139 the reciver parts have been created.
5140 We postpone the creation of the UDP sources until the UDP
5141 ports have been allocated.
5142 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5144 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5146 * gst/rtsp-server/rtsp-stream.c:
5147 rtsp-stream: added function for setting UDP sources to PLAYING state
5148 Code refactoring: Introduced a function for setting UDP sources
5150 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5152 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5154 * gst/rtsp-server/rtsp-stream.c:
5155 rtsp-stream: added function for creating and configuring UDP sources
5156 Code refactoring: create and configure UDP sources in a separate function.
5157 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5159 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5161 * gst/rtsp-server/rtsp-stream.c:
5162 rtsp-stream: added function for RTP/RTCP socket configuration
5163 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5164 in a separate function.
5165 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5167 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5169 * gst/rtsp-server/rtsp-stream.c:
5170 rtsp-stream: added function for creating and configuring UDP sinks
5171 Code refactoring: create and configure UDP sinks in a separate function.
5172 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5174 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5176 * gst/rtsp-server/rtsp-stream.c:
5177 rtsp-stream: added helper function for creating the sender/receiver parts
5178 Code refactoring: introduced helper function for creating
5179 the receiver and the sender parts of the streaming pipeline.
5180 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5182 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5187 === release 1.7.2 ===
5189 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5195 * gst-rtsp-server.doap:
5198 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5200 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5201 uninstalled.pc: add support for non libtool build systems
5202 Currently the .la path is provided which requires to use libtool as
5203 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5204 It is fine as long as the application is built using libtool.
5205 So currently it is not possible to compile a GStreamer application
5206 within gst-uninstalled with CMake or other build system different
5208 This patch allows to do the following in gst-uninstalled env:
5209 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5210 gstreamer-rtsp-server-1.0)
5211 Previously it required to prepend libtool --mode=link
5212 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5214 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5216 * gst/rtsp-sink/gstrtspclientsink.c:
5217 rtspclientsink: remove check for impossible condition
5218 Goto error label checks stream to see if it needs to be unreferenced before
5219 returning, but this goto jumps happens before the stream is ever set, so it
5220 will always be NULL in this error label.
5223 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5225 * gst/rtsp-sink/gstrtspclientsink.c:
5226 rtspclientsink: clean switch statements
5227 Coverity demands for fallthrough statements to be clearly commented,
5228 to distinguish from accidental fall throughs. And it also needs all
5229 cases to finish with a break, even if the break is never going to be
5230 executed like in the case of a continue jump.
5234 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5236 * tests/check/Makefile.am:
5237 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5238 To get the CK_DEFAULT_TIMEOUT defined for all tests
5239 Also removes a 120 seconds timeout that was set as default
5240 explicitly in this module
5241 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5243 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5247 Automatic update of common submodule
5248 From 86e4663 to b64f03f
5250 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5252 * gst/rtsp-server/rtsp-media.c:
5253 rtsp-media: fix state_lock not locked again when preroll fails
5254 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5256 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5259 configure: Move plugin specific flags below all the others
5260 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5261 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5263 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5265 * gst/rtsp-sink/gstrtspclientsink.c:
5266 rtspclientsink: Simplify slightly using new -base API
5267 Use the new Mikey and SDP API in the base plugins libs
5268 to simplify some code.
5269 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5271 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5276 * gst/rtsp-sink/Makefile.am:
5277 * gst/rtsp-sink/gstrtspclientsink.c:
5278 * gst/rtsp-sink/gstrtspclientsink.h:
5279 * gst/rtsp-sink/plugin.c:
5280 * tests/check/Makefile.am:
5281 * tests/check/gst/rtspclientsink.c:
5282 rtspsink: Add rtspclientsink element
5283 Add an rtspclientsink element that accepts streams for which
5284 there is a registered payloader and sends them to
5285 an RTSP server using RECORD.
5286 Sending is synchronised to the pipeline clock. Payload-types
5287 are automatically selected. The 'new-payloader' signal is fired
5288 for custom configuration of payloaders when they are created.
5289 Can now stream a movie like this:
5291 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5292 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5294 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5295 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5296 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5298 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5300 * gst/rtsp-server/rtsp-stream.c:
5301 * gst/rtsp-server/rtsp-stream.h:
5302 rtsp-stream: Add functions for using rtsp-stream from the client
5303 Add a boolean to indicate that the rtsp-stream is running on the
5304 'client' side of an RTSP connection, for sending streams via
5305 RECORD. In that case, the roles of the client/server ports
5306 in transport setup are swapped.
5307 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5309 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5311 * gst/rtsp-server/rtsp-sdp.c:
5312 * gst/rtsp-server/rtsp-sdp.h:
5313 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5314 A new function that adds info from a GstRTSPStream into an SDP message.
5315 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5317 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5319 * gst/rtsp-server/rtsp-media.c:
5320 rtsp-media: Fix mutex beeing unlocked while they should be locked
5321 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5323 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5325 * gst/rtsp-server/rtsp-media-factory.c:
5326 rtsp-media-factory: add missing break in "clock" property setter
5329 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5331 * gst/rtsp-server/rtsp-stream.c:
5332 rtsp-stream: fixed assert during update transport
5333 When RTSP server trying update transport during multicast, it throws an
5334 assert. The assert is thrown because it is trying to get the parent of
5335 an non-existing funnel element.
5336 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5338 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5340 * gst/rtsp-server/rtsp-permissions.h:
5341 * gst/rtsp-server/rtsp-thread-pool.h:
5342 * gst/rtsp-server/rtsp-token.h:
5343 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5344 gtk-doc can handle static inline functions just fine these days,
5345 there's no need for this stuff any more.
5347 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5349 * gst/rtsp-server/rtsp-media.c:
5350 * gst/rtsp-server/rtsp-sdp.c:
5351 sdp: replace duplicated codes to call new base sdp apis
5352 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5354 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5356 * examples/test-netclock.c:
5357 test-netclock: Use the new API to configure a clock directly
5359 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5361 * gst/rtsp-server/rtsp-media-factory.c:
5362 * gst/rtsp-server/rtsp-media-factory.h:
5363 * gst/rtsp-server/rtsp-media.c:
5364 * gst/rtsp-server/rtsp-media.h:
5365 rtsp-media: Add API to directly configure a clock on the media pipelines
5367 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5369 * gst/rtsp-server/rtsp-media.c:
5370 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5372 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5374 * gst/rtsp-server/rtsp-media-factory.c:
5375 rtsp-media-factory: Add FIXME for 2.0
5377 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5379 * gst/rtsp-server/rtsp-stream.c:
5380 rtsp-stream: Fix indentation
5382 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5384 * gst/rtsp-server/rtsp-media.c:
5385 rtsp-media: Do not prepare media after media times out
5386 Deferred calls to start_prepare() can be deferred past the point until
5387 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5388 prepared to wait. Previously there was no lock and no check for this
5389 situation. This meant that a media could be prepared and unprepared
5390 simultaneously by two different threads. Now a lock is in place and a
5391 suitable check is done.
5392 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5394 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5396 * gst/rtsp-server/rtsp-client.c:
5397 * gst/rtsp-server/rtsp-media-factory.c:
5398 * gst/rtsp-server/rtsp-media-factory.h:
5399 * gst/rtsp-server/rtsp-media.c:
5400 * gst/rtsp-server/rtsp-media.h:
5401 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5402 Without TEARDOWN it might be desireable to keep the media running and continue
5403 sending data to the client, even if the RTSP connection itself is
5405 Only do this for session medias that have only UDP transports. If there's at
5406 least on TCP transport, it will stop working and cause problems when the
5407 connection is disconnected.
5408 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5410 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5415 === release 1.7.1 ===
5417 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5423 * gst-rtsp-server.doap:
5426 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5429 configure: Make -Bsymbolic check work with clang.
5430 Update the -Bsymbolic check with the version glib has. This version
5432 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5434 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5436 * gst/rtsp-server/rtsp-session-pool.c:
5437 rtsp-session-pool: Avoid dollar sign ($) in session ids
5438 Live555 in VLC strips off dollar signs and then gets very confused,
5439 we don't loose too much entropy by just skipping it.
5441 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5443 * gst/rtsp-server/rtsp-address-pool.h:
5444 * gst/rtsp-server/rtsp-auth.h:
5445 * gst/rtsp-server/rtsp-client.h:
5446 * gst/rtsp-server/rtsp-media-factory-uri.h:
5447 * gst/rtsp-server/rtsp-media-factory.h:
5448 * gst/rtsp-server/rtsp-media.h:
5449 * gst/rtsp-server/rtsp-mount-points.h:
5450 * gst/rtsp-server/rtsp-permissions.h:
5451 * gst/rtsp-server/rtsp-server.h:
5452 * gst/rtsp-server/rtsp-session-media.h:
5453 * gst/rtsp-server/rtsp-session-pool.h:
5454 * gst/rtsp-server/rtsp-session.h:
5455 * gst/rtsp-server/rtsp-stream-transport.h:
5456 * gst/rtsp-server/rtsp-stream.h:
5457 * gst/rtsp-server/rtsp-thread-pool.h:
5458 * gst/rtsp-server/rtsp-token.h:
5459 rtsp-server: Add g_autoptr() support to all types
5460 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5462 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5464 * gst/rtsp-server/rtsp-stream.c:
5465 rtsp-stream: fixed valgrind error
5466 Fixed the valgrind error in unit test. The UDP source created during
5467 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5469 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5471 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5475 Automatic update of common submodule
5476 From b319909 to 86e4663
5478 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5480 * gst/rtsp-server/rtsp-client.c:
5481 rtsp-client: suspend media during setup request
5482 SETUP request from clients needs to suspend the media to clear the
5483 prerolled buffers. Otherwise it will not affect the prerolled buffer
5484 and the prerolled buffers will be incorrect (for example block-size
5485 from setup request will not affect the prerolled buffer unless the
5486 media is suspended).
5487 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5489 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5491 * gst/rtsp-server/rtsp-stream.c:
5492 rtsp-stream: create stream pipeline based on transport
5493 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5494 only UDP is set as the transport protocol, it will not add the extra tee
5495 or queue element to the pipeline. Both these elements will be added, if
5496 it supports both TCP and UDP protocols. This improves the pipeline
5497 performance when one protocol is present.
5498 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5500 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5502 * gst/rtsp-server/rtsp-stream.c:
5503 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5504 Adding them when not needed will start some logic inside rtpbin that might be
5505 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5506 would start up a rtpjitterbuffer and behave in weird ways.
5507 We still set up the UDP sources for RTP receiving for a sender media to be
5508 able to receive any packets sent by the client for NAT traversal. They will
5509 all go to a fakesink though.
5510 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5511 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5512 receive ASYNC_DONE after a seek.
5513 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5515 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5517 * gst/rtsp-server/rtsp-stream.c:
5518 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5519 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5520 Previously we were only setting this for sender sockets, which caused looped
5521 back packets to be received on Windows if a multicast transport was used.
5523 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5525 * examples/test-record-auth.c:
5526 * examples/test-record.c:
5527 examples: Actually use the provided port in the record examples
5529 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5531 * examples/test-record-auth.c:
5532 test-record-auth: Add the option to build in TLS support
5534 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5536 * examples/test-auth.c:
5537 test-auth: Use an 'anonymous' user for unauthenticated default
5538 There's a comment on one of the resources that 'user' and 'admin'
5539 shouldn't even be able to see it, but they can if the default
5540 token is 'admin2', since that gives them access anyway.
5542 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5544 * examples/.gitignore:
5545 * examples/Makefile.am:
5546 * examples/test-record-auth.c:
5547 Add test-record-auth example
5549 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5551 * gst/rtsp-server/rtsp-client.c:
5552 * tests/check/gst/client.c:
5553 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5555 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5557 * gst/rtsp-server/rtsp-server.c:
5558 rtsp-server: Change the logic so we don't pop a NULL context
5559 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5560 will sometimes fail. This call is made before any context is pushed
5561 resulting in an attempt to pop a NULL context.
5562 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5564 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5566 * tests/check/gst/rtspserver.c:
5567 rtspserver: Add udp-mcast transport SETUP test
5568 Refactor utility functions in the test file so they can handle
5569 more than UDP and TCP as lower transport.
5570 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5572 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5574 * gst/rtsp-server/rtsp-stream.c:
5575 rtsp-stream: Always unref return value of gst_object_get_parent()
5576 Fixes a leak of a GstBin in the udp-mcast case.
5577 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5579 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5582 Automatic update of common submodule
5583 From b99800a to b319909
5585 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5588 Use new GST_ENABLE_EXTRA_CHECKS #define
5589 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5591 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5594 Automatic update of common submodule
5595 From 6babecd to b99800a
5597 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5600 Update GLib dependency to 2.40.0
5602 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5604 * examples/test-mp4.c:
5605 * gst/rtsp-server/rtsp-stream.c:
5606 stream: listen to sender ssrc signals
5607 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5609 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5612 common: update for new suppression
5613 Makes check-valgrind pass with glib 2.46
5615 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5617 * gst/rtsp-server/rtsp-media.c:
5618 rtsp-media: Take reference to media that will be prepared
5619 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5620 Later on a g_idle_source_new() is created and a pointer to the media
5621 object is passed as user data. If the media is freed before the idle
5622 source is dispatched the media object pointer is invalid, but the idle
5623 source callback expects it to still be valid. To fix this a reference to
5624 the media object is taken when registering the source callback function
5625 and a corresponding release of the reference is done when the souce is
5627 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5629 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5631 * examples/test-launch.c:
5632 * examples/test-mp4.c:
5633 * examples/test-ogg.c:
5634 * examples/test-record.c:
5635 * examples/test-uri.c:
5636 rtsp-server: Fix memory leaks when context parse fails
5637 When g_option_context_parse fails, context and error variables are not getting free'd
5638 which results in memory leaks. Free'ing the same.
5639 And replacing g_error_free with g_clear_error, which checks if the error being passed
5640 is not NULL and sets the variable to NULL on free'ing.
5641 https://bugzilla.gnome.org/show_bug.cgi?id=753863
5643 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5648 === release 1.6.0 ===
5650 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5656 * gst-rtsp-server.doap:
5659 === release 1.5.91 ===
5661 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
5667 * gst-rtsp-server.doap:
5670 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
5672 * docs/libs/gst-rtsp-server-sections.txt:
5673 * gst/rtsp-server/rtsp-stream.c:
5674 stream: fix docs for recently-added get/set_buffer_size API
5675 https://bugzilla.gnome.org/show_bug.cgi?id=749095
5677 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
5679 * gst/rtsp-server/rtsp-media.c:
5680 rtsp-media: Don't crash on encrypted RTX SDP
5681 In parse_keymgmt(), don't mutate the input string that's been passed
5682 as const, especially since we might need the original value again if
5683 the same key info applies to multiple streams (RTX, for example).
5684 https://bugzilla.gnome.org/show_bug.cgi?id=754753
5686 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
5688 * examples/test-mp4.c:
5689 test-mp4: Support filenames with spaces in them. Error out on too few arguments
5691 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
5693 * examples/test-record.c:
5694 test-record: Check parameter count and print out help
5695 If no launch pipeline was supplied, print out some help
5697 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
5699 * gst/rtsp-server/rtsp-media.c:
5700 * gst/rtsp-server/rtsp-stream.c:
5701 * gst/rtsp-server/rtsp-stream.h:
5702 rtsp-stream: Implement UDP buffer size setting.
5703 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
5705 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
5706 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
5708 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
5710 * gst/rtsp-server/rtsp-media.h:
5711 rtsp-media: Fix small typo causing gtk-doc to complain
5713 === release 1.5.90 ===
5715 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5721 * gst-rtsp-server.doap:
5724 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5726 * gst/rtsp-server/rtsp-media-factory.c:
5727 media-factory: get port number through gst_rtsp_url_get_port
5728 https://bugzilla.gnome.org/show_bug.cgi?id=753473
5730 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
5732 * tests/check/gst/media.c:
5733 media-test: Removing unnecessary assertion
5734 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5736 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5738 * gst/rtsp-server/rtsp-server.c:
5739 Document that source keeps a ref on server until it's destroyed
5740 https://bugzilla.gnome.org/show_bug.cgi?id=749227
5742 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5744 * tests/check/gst/media.c:
5745 media-test: Test for multiple dynamic payload
5746 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5748 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5750 * gst/rtsp-server/rtsp-media.c:
5751 media: Only add fakesink once per pipeline
5752 The intention is to prevent going PLAYING state before pads are created.
5753 If there was mutilple dynamic payload, it would leak few fakesink and
5754 actually prevent from ever reaching playing state.
5755 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5757 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5759 * gst/rtsp-server/rtsp-media.c:
5760 Revert "rtsp-media: Only add 1 fakesink per pipeline"
5761 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
5763 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5765 * gst/rtsp-server/rtsp-media.c:
5766 rtsp-media: Only add 1 fakesink per pipeline
5767 There should be only one fakesink per pipeline, not per dynpay. This
5768 would lead to element naming clash.
5770 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
5772 * gst/rtsp-server/rtsp-media.c:
5773 rtsp-media: assertion error due to wrong condition check
5774 In media to caps function, reserved_keys array is being used for variable i,
5775 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
5776 changed it to variable j
5777 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5779 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
5781 * gst/rtsp-server/rtsp-media.c:
5782 rtsp-media: Strip keys from the fmtp that we use internally in our caps
5783 Skip keys from the fmtp, which we already use ourselves for the
5784 caps. Some software is adding random things like clock-rate into
5785 the fmtp, and we would otherwise here set a string-typed clock-rate
5786 in the caps... and thus fail to create valid RTP caps
5787 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5789 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5791 * gst/rtsp-server/rtsp-thread-pool.c:
5792 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
5793 https://bugzilla.gnome.org/show_bug.cgi?id=752640
5795 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
5798 Automatic update of common submodule
5799 From f74b2df to 9aed1d7
5801 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
5806 === release 1.5.2 ===
5808 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
5814 * gst-rtsp-server.doap:
5817 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
5819 * gst/rtsp-server/rtsp-client.c:
5820 * gst/rtsp-server/rtsp-client.h:
5821 * tests/check/gst/client.c:
5822 rtsp-client: allow application to decide what requirements are supported
5823 Add "check-requirements" signal and vfunc to allow application
5824 (and subclasses) to check the requirements.
5825 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
5826 https://bugzilla.gnome.org/show_bug.cgi?id=749417
5828 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5831 Automatic update of common submodule
5832 From 6015d26 to f74b2df
5834 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5836 * gst/rtsp-server/rtsp-media.c:
5837 rtsp-media: Always use real payloader when creating streams
5838 A bin that contains the real payloader might be used as payloader. In this
5839 case we have to get the real payloader for the various properties it provides.
5840 Example use cases for this are bins that payload some media and then have
5841 additional elements that add metadata or RTP extension headers to the stream.
5842 https://bugzilla.gnome.org/show_bug.cgi?id=750800
5844 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5846 * examples/test-netclock-client.c:
5847 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
5849 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
5851 * examples/test-netclock-client.c:
5852 * examples/test-netclock.c:
5853 test-netclock: Use new ntp-time-source property on rtpbin
5854 Select the clock time to be used as NTP time source. This allows proper
5855 synchronization between receivers, independent of sharing base times, and just
5856 requires them to use the same clock.
5858 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5860 * examples/test-netclock-client.c:
5861 * examples/test-netclock.c:
5862 test-netclock: Setting the same base time on sender and receiver is not necessary
5863 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
5865 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5867 * gst/rtsp-server/rtsp-stream.c:
5868 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
5869 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5871 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5873 * docs/libs/gst-rtsp-server.types:
5874 docs: add missing types
5875 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5877 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5879 * docs/libs/gst-rtsp-server-sections.txt:
5880 docs: add missing apis
5881 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5883 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5885 * examples/test-netclock-client.c:
5886 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
5888 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5890 * docs/libs/gst-rtsp-server-sections.txt:
5891 * gst/rtsp-server/rtsp-auth.c:
5892 * gst/rtsp-server/rtsp-auth.h:
5893 GstRTSPAuth: Add client certificate authentication support
5894 https://bugzilla.gnome.org/show_bug.cgi?id=750471
5896 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5898 * examples/test-netclock-client.c:
5899 test-netclock-client: Use new GstClock API to wait for clock synchronization
5901 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
5903 * examples/test-netclock-client.c:
5904 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
5905 A mainloop is needed to get glimagesink to display something on OSX, and
5906 the source-setup signal just makes things a little bit easier.
5908 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
5911 Automatic update of common submodule
5912 From d9a3353 to 6015d26
5914 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
5917 Automatic update of common submodule
5918 From d37af32 to d9a3353
5920 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
5923 Automatic update of common submodule
5924 From 21ba2e5 to d37af32
5926 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
5929 Automatic update of common submodule
5930 From c408583 to 21ba2e5
5932 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
5934 * docs/libs/Makefile.am:
5935 docs: remove variables that we define in the snippet from common
5936 This is syncing our Makefile.am with upstream gtkdoc.
5938 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5941 Automatic update of common submodule
5942 From 44a3517 to c408583
5944 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
5949 === release 1.5.1 ===
5951 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
5957 * gst-rtsp-server.doap:
5960 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
5962 * gst/rtsp-server/rtsp-client.c:
5963 rtsp-client: No flush during Teardown.
5964 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
5965 backlog is empty it can happen that just a part of a message will be
5966 sent and rest is in backlog queue. If then flush during teardown
5967 just a part of message will be sent.This can lead to client miss
5968 teardown response since it expect to get the last part of message.
5969 The flushing during teardown was introduced to fix a deadlock that now
5970 is fixed more generally in handle_request by temporary setting backlog
5972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
5974 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
5976 * tests/check/Makefile.am:
5977 tests: Use AM_TESTS_ENVIRONMENT
5978 Needed by the new automake test runner and the
5979 current version of the common submodule.
5981 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5983 * gst/rtsp-server/rtsp-media.h:
5984 * gst/rtsp-server/rtsp-stream.h:
5985 rtsp-server: Use single-include rtsp header to make sure we get all definitions
5987 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
5989 * gst/rtsp-server/rtsp-media.c:
5990 rtsp-media: Mark some more functions static
5992 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
5994 * gst/rtsp-server/rtsp-media.c:
5995 rtsp-media: Only unblock the media in suspend() when actually changing the state
5996 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
5998 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6000 * examples/test-video-rtx.c:
6001 examples: Use AVPF profile for the RTX example
6003 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6005 * gst/rtsp-server/rtsp-sdp.c:
6006 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6008 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6010 * gst/rtsp-server/rtsp-stream.c:
6011 rtsp-stream: get valid clock-rate from last-sample
6012 clock-rate in last-sample's caps is integer, not unsigned.
6013 To get this value properly, variable needs to be type-casted to int.
6014 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6016 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6020 autogen.sh: only run autopoint if gettext requested in configure.ac
6021 Not just because there happens to be a po directory.
6022 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6024 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6027 Revert "configure.ac: uncomment gettext version setup"
6028 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6029 We don't need a gettext setup here and there's no po
6030 directory either, so no reason why autopoint would be
6031 run in the first place.
6032 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6034 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6036 * examples/test-multicast.c:
6037 * examples/test-multicast2.c:
6038 * examples/test-sdp.c:
6039 * examples/test-video-rtx.c:
6040 * examples/test-video.c:
6041 * tests/test-cleanup.c:
6042 * tests/test-reuse.c:
6043 Fix timeout function signatures across tests and examples
6045 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6047 * tests/check/Makefile.am:
6048 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6049 Make sure the test environment is set up.
6050 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6052 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6055 configure: bump automake requirement to 1.14 and autoconf to 2.69
6056 This is only required for builds from git, people can still
6057 build tarballs if they only have older autotools.
6058 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6060 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6063 configure.ac: uncomment gettext version setup
6064 Fixes autogen.sh. It would run autopoint, which would complain
6065 that it could not find the gettext version in configure.ac.
6066 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6068 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6070 * examples/test-video-rtx.c:
6071 test-video-rtx: set exact payload type to PCMA payloader
6072 Setting wrong payload type causes failure to do retransmission through audio stream
6073 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6075 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6077 * gst/rtsp-server/rtsp-media.c:
6078 * gst/rtsp-server/rtsp-stream.c:
6079 * gst/rtsp-server/rtsp-stream.h:
6080 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6081 Because of duplicated g_signal_connect for request-aux-sender signal,
6082 wrong stream pointer is passed to the signal handler.
6083 Instead of passing each stream, pass stream array and get the relevant stream.
6084 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6086 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6090 Update autogen.sh to latest version from common
6091 Fixes build after aclocal_check etc. helpers have been removed.
6093 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6096 Automatic update of common submodule
6097 From bc76a8b to c8fb372
6099 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6101 * gst/rtsp-server/rtsp-stream.c:
6102 rtsp-stream: Limit the queues to 1 buffer
6103 We only need them to be able to pre-roll, queueing up more data here
6104 is only going to harm latency and memory usage.
6106 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6108 * gst/rtsp-server/rtsp-stream.c:
6109 rtsp-stream: Update comment and ASCII art to the latest code
6110 We have a queue in front of the udpsink too to prevent the pipeline from
6113 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6115 * gst/rtsp-server/rtsp-stream.c:
6116 rtsp-media: Properly return first rtptime
6117 Instead we where returning first GstBuffer timestamp. This would result
6118 in clock skew and unwanted behaviour in RTSP playback.
6119 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6121 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6123 * gst/rtsp-server/rtsp-stream.c:
6124 rtsp-stream: Don't leave buffer mapped
6125 If the seq is NULL, the RTP buffer was left mapped. We should always
6128 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6133 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6135 * gst/rtsp-server/rtsp-media-factory.c:
6136 * tests/check/gst/client.c:
6137 Fix double semicolons
6139 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6141 * gst/rtsp-server/rtsp-stream.c:
6142 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6143 This gives more accurate values than asking the payloader. There might be
6144 queueing happening between the payloader and the sink.
6145 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6147 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6149 * gst/rtsp-server/rtsp-media.c:
6150 rtsp-media: Don't seek for PLAY if the position will not change
6151 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6153 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6155 * gst/rtsp-server/rtsp-media.c:
6156 rtsp-media: Don't include payload type in the caps for framesize
6157 When the sdp media attribute framesize are converted to caps
6158 the <payload> should not be included.
6159 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6160 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6162 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6164 * gst/rtsp-server/rtsp-sdp.c:
6165 rtsp-sdp: add payload type to the sdp framesize attribute
6166 The sdp framesize attribute is desribed in RFC6064. It is specified
6167 for payloading of H263 and has the following form
6168 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6169 should be added to the caps in a payloader and the <payload type> should
6170 be added by the rtsp-server.
6171 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6173 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6175 * examples/test-uri.c:
6176 examples: test-uri: fix tainted variable
6177 Insignificant but this keeps Coverity happy.
6180 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6182 * examples/.gitignore:
6183 * examples/Makefile.am:
6184 * examples/test-netclock-client.c:
6185 * examples/test-netclock.c:
6186 examples: Add a simple example of network synch for live streams.
6187 An example server and client that works for synchronising live streams
6188 only - as it can't support pause/play.
6190 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6192 * gst/rtsp-server/rtsp-media-factory.c:
6193 * gst/rtsp-server/rtsp-media-factory.h:
6194 rtsp-media-factory: Add functions to set/get the media gtype
6195 Allow specifying the GType of a GstRtspMedia subclass to create
6196 as a simpler way to get the factory to create a custom
6197 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6199 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6201 * gst/rtsp-server/rtsp-media.c:
6202 rtsp-media: fix double unlock in _get_buffer_size()
6203 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6204 because of double g_mutex_unlock () usage.
6205 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6207 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6209 * gst/rtsp-server/rtsp-session-pool.c:
6210 * gst/rtsp-server/rtsp-session.c:
6211 * gst/rtsp-server/rtsp-session.h:
6212 rtsp-session: Use monotonic time for RTSP session timeout
6213 Changed RTSP session timeout handling to monotonic time
6214 and deprecating the API for current system time.
6215 This fixes timeouts when the system time changes.
6216 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6218 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6220 * gst/rtsp-server/rtsp-client.c:
6221 * gst/rtsp-server/rtsp-media.c:
6222 rtsp-client: Only error out in PLAY if seeking actually failed
6223 If the media was just not seekable, we continue from whatever position we are
6224 and let the client decide if that is what is wanted or not.
6225 Only if the actual seek failed, we can't really recover and should error out.
6227 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6229 * gst/rtsp-server/rtsp-stream.c:
6230 rtsp-stream: Add necessary queues between tee and multiudpsink
6231 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6233 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6235 * gst/rtsp-server/rtsp-client.c:
6236 * gst/rtsp-server/rtsp-media.c:
6237 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6238 Instead error out properly the same way as if the SEEKING query already
6241 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6243 * gst/rtsp-server/rtsp-stream.h:
6244 rtsp-stream: minor code formatting fix
6246 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6248 * gst/rtsp-server/rtsp-media.c:
6249 rtsp-media: fix logic for collect_streams
6250 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6251 all streams it knows if it got any, and can check if the transport mode is OK.
6254 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6256 * gst/rtsp-server/rtsp-media.c:
6257 rtsp-media: Don't set the transport mode based on what elements we find
6258 Just print a warning if the one that was set before disagrees with what
6259 elements we found. It must already be set to something before as this
6260 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6261 and we would reject ANNOUNCE if the RECORD flag was not set.
6263 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6265 * tests/check/gst/rtspserver.c:
6266 tests: rtspserver: rename shadowed variable
6267 We have two different 'sink' variables here,
6268 rename one of them for clarity.
6270 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6272 * gst/rtsp-server/rtsp-client.c:
6273 rtsp-client: fix awkward if clause
6275 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6277 * examples/test-uri.c:
6278 examples: test-uri: improve uri argument handling and accept file names
6279 Print an error if the argument passed is not a URI and can't
6280 be converted into one, or no arguments have been provided.
6282 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6284 * examples/test-uri.c:
6285 examples: test-uri: don't remove mount point after 10 seconds
6286 It's very irritating when trying to test stuff repeatedly
6287 and serves no real purpose other than showing that it can
6290 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6292 * examples/.gitignore:
6293 examples: add new test-record to .gitignore
6295 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6297 * examples/test-record.c:
6298 * gst/rtsp-server/rtsp-client.c:
6299 * gst/rtsp-server/rtsp-media-factory.c:
6300 * gst/rtsp-server/rtsp-media-factory.h:
6301 * gst/rtsp-server/rtsp-media.c:
6302 * gst/rtsp-server/rtsp-media.h:
6303 * tests/check/gst/rtspserver.c:
6304 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6306 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6308 * examples/test-record.c:
6309 test-record: Set latency for playback-style example to 2s instead of 200ms
6311 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6313 * tests/check/gst/rtspserver.c:
6314 tests: add some unit tests for ANNOUNCE and RECORD
6315 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6317 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6319 * gst/rtsp-server/rtsp-client.c:
6320 rtsp-client: fix a couple of leaks in handle_announce
6322 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6324 * gst/rtsp-server/rtsp-media-factory.c:
6325 * gst/rtsp-server/rtsp-media-factory.h:
6326 * gst/rtsp-server/rtsp-media.c:
6327 * gst/rtsp-server/rtsp-media.h:
6328 rtsp-media: Expose latency setting for setting the rtpbin latency
6330 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6332 * examples/test-record.c:
6333 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6335 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6337 * gst/rtsp-server/rtsp-stream.c:
6338 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6340 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6342 * examples/Makefile.am:
6343 * examples/test-record.c:
6344 * gst/rtsp-server/rtsp-client.c:
6345 * gst/rtsp-server/rtsp-client.h:
6346 * gst/rtsp-server/rtsp-media-factory.c:
6347 * gst/rtsp-server/rtsp-media-factory.h:
6348 * gst/rtsp-server/rtsp-media.c:
6349 * gst/rtsp-server/rtsp-media.h:
6350 * gst/rtsp-server/rtsp-session-media.c:
6351 * gst/rtsp-server/rtsp-stream.c:
6352 * gst/rtsp-server/rtsp-stream.h:
6353 Add initial support for RECORD
6354 We currently only support media that is RECORD or PLAY only, not both at once.
6355 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6357 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6359 * gst/rtsp-server/rtsp-stream.c:
6360 rtsp-stream: RTCP and RTP transport cache cookies seperated
6361 RTCP packets were not sent because the same tr_cache_cookie was used for
6362 both RTP and RTCP. So only one of the tr_cache lists were populated
6363 depending on which one was sent first. If the tr_cache list is not
6364 populated then no packets can be sent. Most often this happened to be
6365 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6366 resulted in both the tr_cache_lists to be populated regardless of which
6368 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6370 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6372 * gst/rtsp-server/rtsp-stream.c:
6373 rtsp-stream: fix false compiler warning
6374 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6376 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6378 * gst/rtsp-server/rtsp-client.c:
6379 rtsp-client: log interleaved data received
6381 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6383 * gst/rtsp-server/rtsp-client.c:
6384 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6386 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6388 * gst/rtsp-server/rtsp-client.c:
6389 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6391 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6393 * gst/rtsp-server/rtsp-client.c:
6394 rtsp-client: Use a random session ID in the SDP
6395 RFC4566 Section 5.2 says that it should make the username, session id,
6396 nettype, addrtype and unicast address tuple globally unique. Always using
6397 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6398 Instead let's create a 64 bit random number, which at least brings us
6399 closer to the goal of global uniqueness.
6400 https://tools.ietf.org/html/rfc4566#section-5.2
6402 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6404 * examples/test-launch.c:
6405 * examples/test-mp4.c:
6406 * examples/test-ogg.c:
6407 * examples/test-uri.c:
6408 examples: Don't call gst_init() and gst_get_option_group()
6409 The latter calls the former at the appropriate time.
6411 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6413 * gst/rtsp-server/rtsp-client.c:
6414 rtsp-client: Drop trailing \0 of RTSP DATA messages
6415 We add a trailing \0 in GstRTSPConnection to make parsing of
6416 string message bodies easier (e.g. the SDP from DESCRIBE) but
6417 for actual data this means we have to drop it or otherwise
6418 create invalid data.
6420 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6422 * gst/rtsp-server/rtsp-stream.c:
6423 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6424 Fixes crash when two threads access handle_new_sample() at the same
6425 time, one for RTP, one for RTCP.
6426 Otherwise, when iterating over the transports cache, it might be modified by
6427 another thread at the same time if the transports cookie has changed.
6428 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6430 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6432 * gst/rtsp-server/rtsp-stream.c:
6433 rtsp-stream: Set format=TIME on our app sources for TCP
6435 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6437 * gst/rtsp-server/rtsp-session-pool.c:
6438 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6439 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6440 RFC 2326 states that session IDs may consist of alphanumeric as well as
6441 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6442 Previously the session ID was URI-escaped, this meant that any character
6443 which was not alphanumeric or any of the characters +-._~ would be
6444 percent encoded. While the RFC (surprisingly) mentions that linear white
6445 space in session IDs should be URI-escaped, it does not say anything
6446 about other characters. Moreover no white space is allowed in the
6447 session ID. Finally the percent character which is the result of
6448 URI-escaping is not allowed in a session ID.
6449 So there is no reason to do any URI-escaping, and now it is removed.
6450 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6452 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6455 Automatic update of common submodule
6456 From f2c6b95 to bc76a8b
6458 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6461 Fix 'make check' from top-level directory
6463 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6465 * examples/test-launch.c:
6466 * examples/test-mp4.c:
6467 * examples/test-ogg.c:
6468 * examples/test-uri.c:
6469 examples: Add command-line parsing and take a 'port' argument
6470 This allows users to run multiple servers on different ports for testing.
6471 Only done for examples that actually take arguments and hence are capable of
6472 outputting different streams for each instance on each port.
6473 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6475 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6477 * gst/rtsp-server/rtsp-client.c:
6478 * gst/rtsp-server/rtsp-client.h:
6479 rtsp-client: Add a send_message default signal handler
6480 This allows subclasses to easily hook into the response sending
6481 mechanism without doing everything from a signal, which seems
6482 awkward from subclasses.
6484 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6487 Automatic update of common submodule
6488 From ef1ffdc to f2c6b95
6490 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6494 configure: add --disable-examples switch
6495 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6497 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6499 * examples/.gitignore:
6500 * examples/Makefile.am:
6501 * examples/test-video-rtx.c:
6502 examples: add a retransmisison example implementing RFC4588
6503 Currently only SSRC-multiplexed rtx streams are supported
6505 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6507 * gst/rtsp-server/rtsp-stream.c:
6508 rtsp-stream: Fix some minor memory leaks
6510 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6512 * gst/rtsp-server/rtsp-media.c:
6513 rtsp-media: Some minor cleanup
6515 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6517 * gst/rtsp-server/rtsp-stream.c:
6518 rtsp-stream: Fix compiler warnings
6519 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6520 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6522 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6523 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6526 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6528 * docs/libs/gst-rtsp-server-sections.txt:
6529 * gst/rtsp-server/rtsp-media-factory.c:
6530 * gst/rtsp-server/rtsp-media-factory.h:
6531 * gst/rtsp-server/rtsp-media.c:
6532 * gst/rtsp-server/rtsp-media.h:
6533 * gst/rtsp-server/rtsp-sdp.c:
6534 * gst/rtsp-server/rtsp-stream.c:
6535 * gst/rtsp-server/rtsp-stream.h:
6536 media: implement ssrc-multiplexed retransmission support
6537 based off RFC 4588 and the server-rtpaux example in -good
6539 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6541 * gst/rtsp-server/rtsp-client.c:
6542 * gst/rtsp-server/rtsp-stream-transport.c:
6543 * gst/rtsp-server/rtsp-stream.c:
6544 rtsp: Ref transports in hash table.
6545 Also ref streams for transports.
6546 This solves a crash when reciving a rtcp after teardown but before
6548 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6550 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6553 Automatic update of common submodule
6554 From 7bb2bce to ef1ffdc
6556 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6558 * gst/rtsp-server/rtsp-client.c:
6559 client: refactor cleanup of cached media
6561 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6563 * tests/check/gst/client.c:
6565 The session leak is now fixed, lets remove those FIXME comments.
6567 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6569 * tests/check/gst/rtspserver.c:
6570 tests: Test to setup two sessions on one connection
6571 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6573 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6575 * tests/check/gst/rtspserver.c:
6576 tests: Test setup with tcp transport
6577 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6579 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6581 * gst/rtsp-server/rtsp-client.c:
6582 client: Configure transport after creating session media
6583 The default implementation of configure_client_transport() in
6584 rtsp-client uses the session media when it chooses channels for
6585 interleaved traffic.
6586 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6588 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6590 * gst/rtsp-server/rtsp-client.c:
6591 * gst/rtsp-server/rtsp-session-media.c:
6592 client: Stop caching media in client when doing setup
6593 If the media has been managed by a session media, it should not be
6594 cached in the client any longer. The GstRTSPSessionMedia object is now
6595 responsible for unpreparing the GstRTSPMedia object using
6596 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6598 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6600 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6602 * gst/rtsp-server/rtsp-stream.c:
6603 rtsp-stream: unref srtp decoder when leaving bin
6604 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6606 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6608 * gst/rtsp-server/rtsp-client.c:
6609 rtsp-client: mikey memory leaks
6610 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6612 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6615 Automatic update of common submodule
6616 From 84d06cd to 7bb2bce
6618 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6621 Parallelise 'make check-valgrind'
6623 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6626 Automatic update of common submodule
6627 From a8c8939 to 84d06cd
6629 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6632 Automatic update of common submodule
6633 From 36388a1 to a8c8939
6635 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6637 * gst/rtsp-server/rtsp-media.c:
6638 rtsp-media: deactivate media when shutting down from paused
6639 This was only done when going directly from playing.
6640 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
6642 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6644 * gst/rtsp-server/rtsp-client.c:
6645 * gst/rtsp-server/rtsp-context.h:
6646 rtsp-client: add stream transport to context
6647 We add the stream transport to the context so we can get the configured
6648 client stream transport in the setup request signal.
6649 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
6651 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6653 * gst/rtsp-server/rtsp-stream.c:
6654 stream: release lock even not all transports have been removed
6655 We don't want to keep the lock even we return FALSE because not all the
6656 transports have been removed. This could lead into a deadlock.
6657 https://bugzilla.gnome.org/show_bug.cgi?id=737797
6659 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
6661 * gst/rtsp-server/rtsp-sdp.c:
6662 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
6663 These were renamed in GstRTPBasePayload in 1.0
6665 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6667 * gst/rtsp-server/rtsp-client.c:
6668 client: set session media to NULL without the lock
6669 We need to set session medias to NULL without the client lock otherwise
6670 we can end up in a deadlock if another thread is waiting for the lock
6671 and media unprepare is also waiting for that thread to end.
6672 https://bugzilla.gnome.org/show_bug.cgi?id=737690
6674 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
6676 * gst/rtsp-server/rtsp-media.c:
6677 rtsp-media: Set state to UNPREPARING in all cases
6679 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
6681 * gst/rtsp-server/rtsp-media.c:
6682 media: set state to unpreparing when unprepare is initiated
6683 https://bugzilla.gnome.org/show_bug.cgi?id=737675
6685 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
6687 * gst/rtsp-server/rtsp-client.c:
6688 rtsp-client: Remove backlog limit while processings requests
6689 If the backlog limit is kept two cases of deadlocks may be
6690 encountered when streaming over TCP. Without the backlog
6691 limit this deadlocks can not happen, at the expence of
6693 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
6695 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
6697 * gst/rtsp-server/rtsp-client.c:
6698 rtsp-client: do not free main context before rtsp watch
6699 https://bugzilla.gnome.org/show_bug.cgi?id=737110
6701 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
6703 * tests/check/gst/rtspserver.c:
6704 tests: Extend unit test timeout to accomodate for valgrind
6705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6707 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
6709 * gst/rtsp-server/rtsp-client.c:
6710 * gst/rtsp-server/rtsp-session.c:
6711 * gst/rtsp-server/rtsp-stream-transport.c:
6712 rtsp-*: Treat sending packets to clients as keepalive
6713 As long as gst-rtsp-server can successfully send RTP/RTCP data to
6714 clients then the client must be reading. This change makes the server
6715 timeout the connection if the client stops reading.
6716 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6718 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
6720 * gst/rtsp-server/rtsp-client.c:
6721 rtsp-client: Allow backlog to grow while expiring session
6722 Allow the send backlog in the RTSP watch to grow to unlimited size while
6723 attempting to bring the media pipeline to NULL due to a session
6724 expiring. Without this change the appsink element cannot change state
6725 because it is blocked while rendering data in the new_sample callback.
6726 This callback will block until it has successfully put the data into the
6727 send backlog. There is a chance that the send backlog is full at this
6728 point which means that the callback may block for a long time, possibly
6729 forever. Therefore the media pipeline may also be prevented from
6730 changing state for a long time.
6731 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6733 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
6735 * gst/rtsp-server/rtsp-client.c:
6736 rtsp-client: Make old compilers happy
6737 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
6738 Just in case that guint8 doesn't fit in a pointer. Just in case ...
6740 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
6742 * gst/rtsp-server/rtsp-client.c:
6743 client: raise the backlog limits before pausing
6744 We need to raise the backlog limits before pausing the pipeline or else
6745 the appsink might be blocking in the render method in wait_backlog() and
6746 we would deadlock waiting for paused.
6747 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
6749 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
6751 * gst/rtsp-server/rtsp-client.c:
6752 client: make define for the WATCH_BACKLOG
6753 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
6755 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
6757 * gst/rtsp-server/rtsp-client.c:
6758 client: simplify session transport handling
6759 link/unlink of the transport in a session was done to keep track of all
6760 TCP transports and to send RTP/RTCP data to the streams. We can simplify
6761 that by putting all the TCP transports in a hashtable indexed with the
6763 We also don't need to link/unlink the transports when we pause/resume
6764 the streams. The same effect is already achieved when we pause/play the
6765 media. Indeed, when we pause the media, the transport is removed from
6766 the media and the callbacks will not be called anymore.
6767 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
6769 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
6771 * gst/rtsp-server/rtsp-stream-transport.c:
6772 * gst/rtsp-server/rtsp-stream-transport.h:
6773 stream-transport: make method to handle received data
6774 Make a method to handle the data received on a channel. It sends the
6775 data to the stream of the transport on the RTP or RTCP pads based on
6778 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
6780 * examples/test-mp4.c:
6781 test: add example of dumping RTCP reports
6783 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
6785 * gst/rtsp-server/rtsp-media.c:
6786 * gst/rtsp-server/rtsp-stream.c:
6787 * gst/rtsp-server/rtsp-stream.h:
6788 rtsp-media: Make sure that sequence numbers are monotonic after pause
6789 The sequence number is not monotonic for RTP packets after pause. The
6790 reason is basepayloader generates a randon sequence number when the
6791 pipeline goes from ready to pause. With this fix generation of sequence
6792 number will be monotonic when going from pause to play request.
6793 https://bugzilla.gnome.org/show_bug.cgi?id=736017
6795 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
6797 * gst/rtsp-server/rtsp-client.c:
6798 rtsp-client: Protect saved clients watch with a mutex
6799 Fixes a crash when close() is called while merging clients
6800 in handle_tunnel(). In that case close() would destroy the
6801 watch while it is still being used in handle_tunnel().
6802 https://bugzilla.gnome.org/show_bug.cgi?id=735570
6804 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
6806 * gst/rtsp-server/rtsp-stream.c:
6807 rtsp-stream: Remove the multicast group udp sources when removing from the bin
6809 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6811 * gst/rtsp-server/rtsp-media.c:
6812 * gst/rtsp-server/rtsp-stream.c:
6813 * gst/rtsp-server/rtsp-stream.h:
6814 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
6815 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
6816 seeking and will always continue counting the time. This leads to
6817 the NPT after a backwards seek to be something completely different
6818 to the actual seek position.
6819 https://bugzilla.gnome.org/show_bug.cgi?id=732644
6821 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
6823 * examples/test-appsrc.c:
6824 examples: fix another reference leak
6825 gst_rtsp_media_get_element() returns a new ref.
6827 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6829 * examples/test-appsrc.c:
6830 examples: unref element after usage
6831 gst_bin_get_by_name_recurse_up() returns an element
6832 reference that must be unreffed after usage.
6833 https://bugzilla.gnome.org/show_bug.cgi?id=734546
6835 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
6837 * gst/rtsp-server/rtsp-media.c:
6838 signals: Fix copy-pasto in target-state signal offset
6840 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
6844 Makefile: Add usage of build-checks step
6845 Allows building checks without running them
6847 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
6849 * gst/rtsp-server/rtsp-stream.c:
6850 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
6851 When a UDP multicast transport is used it is expected that the server listens
6852 for RTP and RTCP packets on the multicast group with the corresponding port.
6853 Without this we will never get RTCP packets from clients in multicast mode.
6854 https://bugzilla.gnome.org/show_bug.cgi?id=732238
6856 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6861 === release 1.4.0 ===
6863 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6869 * gst-rtsp-server.doap:
6872 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
6874 * gst/rtsp-server/rtsp-media.h:
6875 media: correct misspelled words in description
6876 https://bugzilla.gnome.org/show_bug.cgi?id=733244
6878 === release 1.3.91 ===
6880 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6886 * gst-rtsp-server.doap:
6889 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
6891 * docs/libs/gst-rtsp-server-sections.txt:
6894 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
6896 * gst/rtsp-server/rtsp-server.c:
6897 server: implement client REMOVE filter
6899 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
6901 * gst/rtsp-server/rtsp-client.c:
6902 * gst/rtsp-server/rtsp-client.h:
6903 client: expose _close() method
6904 Expose a previously internal close method to close the client
6907 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
6909 * gst/rtsp-server/rtsp-session-pool.c:
6910 session-pool: signal session-removed outside of the lock
6911 Release the lock before emiting the session-removed signal.
6913 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
6915 * gst/rtsp-server/rtsp-client.c:
6916 * gst/rtsp-server/rtsp-server.c:
6917 * gst/rtsp-server/rtsp-session-pool.c:
6918 * gst/rtsp-server/rtsp-session.c:
6919 * gst/rtsp-server/rtsp-stream.c:
6920 filter: Release lock in filter functions
6921 Release the object lock before calling the filter functions. We need to
6922 keep a cookie to detect when the list changed during the filter
6923 callback. We also keep a hashtable to make sure we only call the filter
6924 function once for each object in case of concurrent modification.
6925 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
6927 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
6929 * gst/rtsp-server/rtsp-client.c:
6930 client: check if watch is set in handle_teardown()
6931 The unit tests run without a watch
6933 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6935 * tests/check/gst/client.c:
6936 client tests: send teardown to cleanup session
6938 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
6940 * tests/check/gst/rtspserver.c:
6941 server tests: send teardown to cleanup session
6943 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6945 * gst/rtsp-server/rtsp-client.c:
6946 client: keep ref to client for the session removed handler
6947 This extra ref will be dropped when all client sessions have been
6948 removed. A session is removed when a client sends teardown, closes its
6949 endpoint of the TCP connection or the sessions expires.
6950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6952 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
6954 * gst/rtsp-server/rtsp-client.c:
6955 * gst/rtsp-server/rtsp-session.c:
6956 * tests/check/gst/client.c:
6957 client: manage media in session as a last step
6958 Once we manage a media in a session, we can't unmanage it anymore
6959 without destroying it. Therefore, first check everything before we
6960 manage the media, otherwise if something is wrong we have no way to
6962 If we created a new session and something went wrong, remove the session
6963 again. Fixes a leak in the unit test.
6965 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
6967 * examples/test-mp4.c:
6968 * examples/test-ogg.c:
6969 examples: print 'stream ready at url' for mp4 and ogg example
6971 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
6973 * gst/rtsp-server/rtsp-client.c:
6974 * gst/rtsp-server/rtsp-sdp.c:
6975 rtsp: fix for MIKEY api change
6977 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
6979 * gst/rtsp-server/rtsp-client.c:
6980 client: free watch context only once
6981 The watch context is freed when the source is destroyed. Avoids
6982 a CRITICAL when we try to unref the context twice.
6984 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
6986 * gst/rtsp-server/rtsp-client.c:
6989 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
6991 * gst/rtsp-server/rtsp-client.c:
6992 client: protect sessions with lock
6993 Protect the list of sessions with the lock.
6994 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6996 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
6998 * gst/rtsp-server/rtsp-client.c:
6999 Client: keep a ref to the session
7000 Don't just keep a weak ref to the session objects but use a hard ref. We
7001 will be notified when a session is removed from the pool (expired) with
7002 the new session-removed signal.
7003 Don't automatically close the RTSP connection when all the sessions of
7004 a client are removed, a client can continue to operate and it can create
7005 a new session if it wants. If you want to remove the client from the
7006 server, you have to use gst_rtsp_server_client_filter() now.
7007 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7008 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7010 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7012 * gst/rtsp-server/rtsp-session-pool.c:
7013 * gst/rtsp-server/rtsp-session-pool.h:
7014 session-pool: add session-removed signal
7015 Add a signal to be notified when a session is removed from the pool.
7017 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7019 * gst/rtsp-server/Makefile.am:
7020 * gst/rtsp-server/rtsp-server.h:
7021 Make rtsp-server.h a single-include header, use it for G-I
7022 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7024 === release 1.3.90 ===
7026 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7032 * gst-rtsp-server.doap:
7035 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7037 * gst/rtsp-server/rtsp-stream.c:
7038 stream: crypto can be NULL
7040 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7042 * gst/rtsp-server/rtsp-client.c:
7043 * gst/rtsp-server/rtsp-media.c:
7044 * gst/rtsp-server/rtsp-mount-points.c:
7045 introspection: add missing allow-none annotations
7046 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7048 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7050 * gst/rtsp-server/rtsp-address-pool.c:
7051 * gst/rtsp-server/rtsp-media.c:
7052 * gst/rtsp-server/rtsp-session-media.c:
7053 * gst/rtsp-server/rtsp-session-pool.c:
7054 * gst/rtsp-server/rtsp-stream-transport.c:
7055 * gst/rtsp-server/rtsp-stream.c:
7056 * gst/rtsp-server/rtsp-token.c:
7057 introspection: add (nullable) annotations to return values
7058 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7060 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7062 * gst/rtsp-server/rtsp-client.c:
7063 * gst/rtsp-server/rtsp-stream.c:
7064 gi: improve annotations
7065 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7067 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7069 * gst/rtsp-server/rtsp-client.c:
7070 * gst/rtsp-server/rtsp-media-factory.c:
7071 * gst/rtsp-server/rtsp-media.c:
7072 * gst/rtsp-server/rtsp-server.c:
7073 signals: use generic marshal function
7074 Use the generic C marshal function.
7075 Use more explicit type instead of G_TYPE_POINTER
7077 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7079 * gst/rtsp-server/rtsp-context.h:
7080 context: add type macro
7082 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7084 * gst/rtsp-server/rtsp-client.c:
7085 * gst/rtsp-server/rtsp-sdp.c:
7086 * gst/rtsp-server/rtsp-sdp.h:
7087 sdp: hide key length defines
7088 They don't have a namespace.
7090 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7095 === release 1.3.3 ===
7097 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7103 * gst-rtsp-server.doap:
7106 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7108 * gst/rtsp-server/rtsp-client.c:
7109 * gst/rtsp-server/rtsp-sdp.c:
7110 * gst/rtsp-server/rtsp-sdp.h:
7111 mikey: add different key length parameters
7112 Add encryption and authentication key length parameters to MIKEY. For
7113 the encoders, the key lengths are obtained from the cipher and auth
7114 algorithms set in the caps. For the decoders, they are obtained while
7115 parsing the key management from the client.
7116 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7118 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7120 * tests/check/gst/stream.c:
7121 stream tests: Make sure we get right multicast address from stream
7122 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7124 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7126 * gst/rtsp-server/rtsp-client.c:
7127 client: ref the context until rtsp watch is alive
7128 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7130 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7132 * gst/rtsp-server/rtsp-client.c:
7133 client: Destroy the rtsp watch after connection close
7135 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7137 * gst/rtsp-server/rtsp-media.c:
7138 media: fix confusing comment
7140 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7142 * gst/rtsp-server/rtsp-session.c:
7143 rtsp-session: Timeout in header.
7144 Adding the possbilty to always have timout in header.
7145 This is configurabe with setting "timeout-always-visible".
7146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7148 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7153 === release 1.3.2 ===
7155 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7162 * gst-rtsp-server.doap:
7165 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7168 Automatic update of common submodule
7169 From 211fa5f to 1f5d3c3
7171 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7173 * gst/rtsp-server/rtsp-client.c:
7174 client: store TCP ports in transport
7175 Store the TCP ports in the transport when we are doing RTSP over TCP.
7176 This way, we can easily get to the ports from the transport.
7177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7179 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7181 * gst/rtsp-server/rtsp-stream.c:
7182 stream: add signals for new RTP/RTCP encoders
7183 New signals to allow the user to configure the dynamically created
7185 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7187 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7189 * gst/rtsp-server/rtsp-media.c:
7190 * gst/rtsp-server/rtsp-media.h:
7191 media: Make suspend()/unsuspend() virtual
7192 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7194 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7196 * gst/rtsp-server/rtsp-client.c:
7197 client: fix send-message signal marshaller
7198 Use generic marshalling for the send-message signal. It has
7199 two POINTER arguments, not just one.
7200 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7202 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7204 * tests/check/gst/media.c:
7205 tests: add and remove pads only once
7206 In this test we simulate a dynamic pad by watching the caps event.
7207 Because of renegotiation in the base payloader now, this caps is sent
7208 multiple times but we can only deal with 1 invocation, use a variable to
7209 only 'add and remove' the pad once.
7211 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7213 * tests/check/gst/rtspserver.c:
7214 tests: add unit test for correct handling of Require headers
7215 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7217 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7219 * gst/rtsp-server/rtsp-client.c:
7220 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7221 Servers must handle Require headers and must report a failure
7222 if they don't handle any of the Required options, see RFC 2326,
7223 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7224 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7226 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7231 === release 1.3.1 ===
7233 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7239 * gst-rtsp-server.doap:
7242 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7245 Automatic update of common submodule
7246 From bcb1518 to 211fa5f
7248 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7253 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7255 * tests/check/gst/sessionmedia.c:
7256 tests: fix memory leak in sessionmedia unit test
7258 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7260 * gst/rtsp-server/rtsp-client.c:
7261 client: emit a signal before sending a message
7262 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7264 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7266 * gst/rtsp-server/rtsp-client.c:
7267 client: pass context to send_message
7268 Pass the current context to send_message, we will need it later.
7270 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7272 * gst/rtsp-server/rtsp-client.c:
7273 client: fix typo in comment
7275 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7277 * gst/rtsp-server/rtsp-media.c:
7278 media: Do not stop thread twice if default_prepare() fails
7280 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7282 * gst/rtsp-server/rtsp-client.c:
7283 client: set the watch to flushing before going to NULL
7284 First set the watch to flushing so that we unblock any current and
7285 future attempt to send data on the watch, Then set the pipeline to
7287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7289 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7291 * gst/rtsp-server/rtsp-session-pool.c:
7292 * tests/check/gst/sessionpool.c:
7293 rtsp-session-pool: Fixes annotation
7294 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7295 in the sessionpool test.
7296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7298 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7300 * gst/rtsp-server/rtsp-media.c:
7301 * gst/rtsp-server/rtsp-media.h:
7302 media: make media_prepare virtual
7303 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7305 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7307 * gst/rtsp-server/rtsp-media.c:
7308 * tests/check/gst/media.c:
7309 media: stop the thread in more error cases
7311 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7313 * gst/rtsp-server/rtsp-media.c:
7314 * tests/check/gst/media.c:
7315 media: allow NULL as the thread
7316 Use the default context whan passing a NULL thread.
7318 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7320 * gst/rtsp-server/rtsp-client.c:
7321 rtsp-client: indent cleanup
7322 Coverity was moaning about unreachable code, and I think it was just
7323 confused by { being before the label. We'll see if it pops up again.
7326 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7328 * gst/rtsp-server/rtsp-client.c:
7329 * gst/rtsp-server/rtsp-media.c:
7330 client: Add drop-backlog property
7331 When we have too many messages queued for a client (currently hardcoded
7332 to 100) we overflow and drop the messages. Add a drop-backlog property
7333 to control this behaviour. Setting this property to FALSE will retry
7334 to send the messages to the client by waiting for more room in the
7336 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7338 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7340 * gst/rtsp-server/rtsp-client.c:
7341 client: support for POST before GET when setting up a tunnel
7343 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7345 * gst/rtsp-server/rtsp-client.c:
7346 client: remove watch of the second client after http tunnel setup
7347 The second client will be freed after the HTTP tunnel has been set up.
7348 Make sure it's RTSP watch is never dispatched again.
7349 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7351 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7353 * gst/rtsp-server/rtsp-media.c:
7354 * tests/check/gst/media.c:
7355 media: Make media_prepare() fail if port allocation fails
7356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7358 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7360 * tests/check/gst/media.c:
7361 media test: cleanup the thread pool in tests
7363 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7365 * gst/rtsp-server/rtsp-media.c:
7366 * tests/check/gst/media.c:
7367 rtsp-media: Unblock blocked streams in unprepare
7368 The streams will be blocked when a live media is prepared.
7369 The streams should be unblocked in gst_rtsp_media_unprepare.
7370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7372 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7374 * gst/rtsp-server/rtsp-media.c:
7375 media: release the state lock when going to NULL
7376 Set our state to UNPREPARING and release the state-lock before
7377 setting the pipeline to the NULL state. This way, any pad-added
7378 callback will be able to take the state-lock and check that we are now
7379 unpreparing instead of deadlocking.
7380 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7382 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7384 * gst/rtsp-server/rtsp-media.c:
7385 media: protect status with lock
7386 Make sure we only update the status with the lock.
7388 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7390 * gst/rtsp-server/rtsp-client.c:
7391 * gst/rtsp-server/rtsp-sdp.c:
7392 rtsp: update for MIKEY API changes
7394 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7396 * gst/rtsp-server/rtsp-client.c:
7397 client: parse the mikey response from the client
7398 Parse the mikey response from the client and update the policy for
7401 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7403 * gst/rtsp-server/rtsp-stream.c:
7404 * gst/rtsp-server/rtsp-stream.h:
7405 stream: add method to set crypto info
7406 Make a method to configure the crypto information of a stream.
7407 Set udpsrc in READY instead of PAUSED so that we can configure caps
7410 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7412 * gst/rtsp-server/rtsp-client.c:
7413 client: cleanup error paths
7415 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7417 * gst/rtsp-server/rtsp-media.c:
7420 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7422 * examples/test-video.c:
7423 test: enable SRTP only on RTSPS
7424 We only want to enable SRTP when doing rtsp over TLS so that we can
7425 exchange the keys in a secure way.
7427 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7429 * examples/test-video.c:
7430 test: print an error on failure
7432 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7435 * examples/test-video.c:
7436 * gst/rtsp-server/rtsp-sdp.c:
7437 * gst/rtsp-server/rtsp-stream.c:
7438 * tests/check/Makefile.am:
7439 stream: add SRTP support
7440 Install srtp encoder and decoder elements in rtpbin
7443 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7445 * tests/check/Makefile.am:
7446 * tests/check/gst/sessionpool.c:
7447 tests: Add unit tests for sessionpool
7448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7450 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7452 * tests/check/gst/threadpool.c:
7453 tests: Improve code coverage of rtsp-threadpool tests
7454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7456 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7458 * tests/check/gst/sessionmedia.c:
7459 tests: Improve code coverage for rtsp-session-media
7460 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7462 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7464 gobject-introspection: Add annotations to support language bindings
7465 In addition a few cosmetic changes:
7466 * Adjust the order of arguments
7467 * Fix typo: occured -> occurred
7468 * Fix indentation after Return:-clauses
7469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7471 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7473 * gst/rtsp-server/rtsp-stream.c:
7474 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7475 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7477 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7479 * gst/rtsp-server/rtsp-stream.c:
7480 stream: take caps after the session manager
7481 Take the caps for the SDP after they leave the rtpbin so that we can
7482 also get the properties added by rtpbin elements.
7484 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7486 * gst/rtsp-server/rtsp-stream.c:
7487 stream: release lock while pushing out packets
7488 Keep a cache of the transports and use this to iterate the transport
7489 while pushing packets. This allows us to release the lock early.
7490 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7492 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7494 * gst/rtsp-server/rtsp-client.c:
7495 * gst/rtsp-server/rtsp-client.h:
7496 rtsp-client: vmethod for modifying tunnel GET response
7497 Add a vmethod tunnel_http_response where the response to the HTTP GET
7498 for tunneled connections can be modified.
7499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7501 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7503 * gst/rtsp-server/rtsp-sdp.c:
7504 sdp: make 1 media line per profile
7505 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7506 line in the SDP for each profile. The client is then supposed to pick
7507 one of the profiles in the SETUP request. Because the m= lines have the
7508 same pt, the client also knows that only 1 option is possible.
7510 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7512 * gst/rtsp-server/rtsp-media-factory.c:
7513 * gst/rtsp-server/rtsp-media-factory.h:
7514 * gst/rtsp-server/rtsp-media.c:
7515 factory: add profile property and pass to media and streams
7517 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7519 * examples/test-multicast.c:
7520 * gst/rtsp-server/rtsp-sdp.c:
7521 sdp: pass multicast connection for multicast-only stream
7522 Pass the multicast address of the stream in the connection info in the
7523 SDP so that clients try a multicast connection first.
7524 Only allow multicast connections in the test-multicast example. Also
7525 increase the TTL a little.
7527 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7530 .gitignore: Ignore gcov intermediate files
7531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7533 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7535 * gst/rtsp-server/rtsp-stream.c:
7536 stream: release some locks in error cases
7538 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7540 docs: Enable and fix gtk-doc warnings
7541 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7542 * addresspool/mediafactory: Add missing annotation colon
7543 * stream: Annotate return value
7544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7546 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7549 Automatic update of common submodule
7550 From fe1672e to bcb1518
7552 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7555 Automatic update of common submodule
7556 From 1a07da9 to fe1672e
7558 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7560 * examples/Makefile.am:
7561 examples: use LDADD for libs instead of LDFLAGS
7563 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7566 configure: make sure releases are in .doap file
7568 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7570 * examples/test-cgroups.c:
7571 examples: test-cgroups: don't put code with side effects into g_assert()
7572 The g_assert() might get compiled out with the right
7573 compiler/preprocessor flags.
7575 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7577 * examples/.gitignore:
7578 examples: add cgroup test binary to .gitignore
7580 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7582 * examples/test-cgroups.c:
7583 examples: fix cgroup test build
7584 Fixes build failure caused by compiler warning:
7585 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7587 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7590 .gitignore: ignore temp files created in the course of 'make check'
7592 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7594 * gst/rtsp-server/rtsp-media.c:
7595 rtsp-media: don't loose frames handling new PLAY request
7596 If client supplied a range check if the range specifies the start point.
7597 If not, then do an accurate seek to the current position. If a start
7598 point was specified do do a key unit seek to make sure the streaming
7599 starts with decodeable frames.
7600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7602 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7604 * gst/rtsp-server/rtsp-media.c:
7605 Revert "media: only flush when setting a new start position"
7606 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7607 We need to do the flush in all cases, demuxer block currently for
7610 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7612 * gst/rtsp-server/rtsp-media.c:
7613 media: only flush when setting a new start position
7614 Only flush the pipeline when we change the start position with
7616 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7618 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7620 * gst/rtsp-server/rtsp-stream.c:
7621 stream: set ttl-mc before adding the socket
7622 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7623 never be set on socket.
7624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7626 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7628 * gst/rtsp-server/rtsp-media.c:
7629 media: stop thread if media is already prepared
7630 in gst_rtsp_media_prepare() the thread is not used if media is already
7631 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7633 https://bugzilla.gnome.org/show_bug.cgi?id=724182
7635 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
7638 build: Ship gst-rtsp-server.doap file
7640 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
7642 * tests/check/gst/rtspserver.c:
7643 tests: Fix another compiler warning with gcc
7645 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
7647 * gst/rtsp-server/rtsp-client.c:
7648 * gst/rtsp-server/rtsp-mount-points.c:
7649 * gst/rtsp-server/rtsp-stream.c:
7650 * tests/check/gst/client.c:
7651 rtsp-server: Fix lots of compiler warnings with clang
7653 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
7656 * gst-rtsp-server.doap:
7657 * tests/Makefile.am:
7658 configure: Synchronise with the configure scripts of the other modules
7660 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
7663 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
7665 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
7667 * gst/rtsp-server/rtsp-media.c:
7668 * gst/rtsp-server/rtsp-stream.c:
7669 Revert "rtsp-server: support build against last stable release"
7670 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
7671 Let us require 1.2.3 now, which is going to be released in a few
7674 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
7676 * gst/rtsp-server/rtsp-session-media.c:
7677 * gst/rtsp-server/rtsp-stream-transport.c:
7678 session: improve RTP-Info
7679 Ignore streams that can't generate RTP-Info instead of failing.
7680 Don't return the empty string when all streams are unconfigured but
7681 return NULL so that we don't generate and empty RTP-Info header.
7682 Improve docs a little.
7684 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
7686 * gst/rtsp-server/rtsp-session-media.c:
7687 Don't free rtpinfo GString when it is NULL
7688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7690 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
7692 * gst/rtsp-server/rtsp-media.c:
7693 media: only set keyframe flag when modifying start
7694 Only set the keyframe flag when we modify the start position. The
7695 keyframe flag should probably be ignored when no change is requested but
7696 until we can claim this is all documented properly and all demuxer
7697 implement this, avoid setting the flag.
7698 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
7700 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
7702 * gst/rtsp-server/rtsp-thread-pool.c:
7703 thread-pool: Unref source after mainloop has quit to avoid races in GLib
7704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
7706 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
7708 * gst/rtsp-server/rtsp-stream.c:
7709 stream: handle NULL seqnum and rtptime arguments
7711 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
7713 * gst/rtsp-server/rtsp-thread-pool.c:
7714 * tests/check/gst/threadpool.c:
7715 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
7716 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
7718 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
7720 * gst/rtsp-server/rtsp-stream.c:
7721 stream: add fallback for missing stats property
7722 Use a fallback when the payloader does not have a stats property
7723 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7725 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
7728 Automatic update of common submodule
7729 From f7bc1c3 to 1a07da9
7731 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
7733 * gst/rtsp-server/rtsp-stream.c:
7734 stream: don't leak stats structure
7735 Don't leak the stats structure and deal with NULL stats.
7737 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
7739 * gst/rtsp-server/rtsp-stream.c:
7740 stream: Get rtpinfo properties atomically from payloader
7741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
7743 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
7745 * gst/rtsp-server/rtsp-media.c:
7746 media: refactor state change functions and signals
7747 Make functions to set the target state and the pipeline state and emit
7748 the signals from those functions.
7750 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
7752 * gst/rtsp-server/rtsp-media.c:
7753 * gst/rtsp-server/rtsp-media.h:
7754 media: add signal to notify of pending state changes
7756 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
7758 * gst/rtsp-server/rtsp-media.c:
7759 * gst/rtsp-server/rtsp-stream.c:
7760 rtsp-server: support build against last stable release
7761 Until 1.2.3 is out with the new get_type function and we
7764 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
7766 * gst/rtsp-server/rtsp-stream.c:
7767 stream: fix compilation
7769 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
7771 * gst/rtsp-server/rtsp-media.c:
7772 * gst/rtsp-server/rtsp-media.h:
7773 * gst/rtsp-server/rtsp-stream.c:
7774 * gst/rtsp-server/rtsp-stream.h:
7775 stream: add property to configure profiles
7777 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
7779 * gst/rtsp-server/rtsp-client.c:
7780 client: let stream check supported transport
7781 Delegate the check if a transport is allowed to the stream.
7782 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
7784 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
7786 * gst/rtsp-server/rtsp-stream.c:
7787 * gst/rtsp-server/rtsp-stream.h:
7788 stream: add method to check supported transport
7789 Add a method to check if a transport is supported
7791 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
7794 configure.ac: Only check for gstreamer-check, not check
7795 We include check in gstreamer-check since quite some time now.
7797 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
7799 * gst/rtsp-server/rtsp-session-media.c:
7800 * gst/rtsp-server/rtsp-stream-transport.c:
7801 * gst/rtsp-server/rtsp-stream.c:
7802 * gst/rtsp-server/rtsp-stream.h:
7803 stream: return clock-rate from get_rtpinfo
7804 And use it to correct the rtptime to the requested start-time.
7805 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
7807 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
7809 * gst/rtsp-server/rtsp-session-media.c:
7810 * gst/rtsp-server/rtsp-stream-transport.c:
7811 * gst/rtsp-server/rtsp-stream-transport.h:
7812 session-media: calculate start-time
7814 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
7816 * gst/rtsp-server/rtsp-stream-transport.c:
7817 * gst/rtsp-server/rtsp-stream.c:
7818 * gst/rtsp-server/rtsp-stream.h:
7819 stream: also return the running-time
7820 Return the running-time in the rtpinfo as well.
7822 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
7824 * gst/rtsp-server/rtsp-client.c:
7825 * gst/rtsp-server/rtsp-session-media.c:
7826 * gst/rtsp-server/rtsp-session-media.h:
7827 * gst/rtsp-server/rtsp-stream-transport.c:
7828 * gst/rtsp-server/rtsp-stream-transport.h:
7829 session-media: let the session-media make the RTPInfo
7830 Add method to create the RTPInfo for a stream-transport.
7831 Add method to create the RTPInfo for all stream-transports in a
7833 Use the session-media RTPInfo code in client. This allows us to refactor
7834 another method to link the TCP callbacks.
7836 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7838 mount-points: sort sequence before g_sequence_lookup
7839 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
7840 sort sequence if dirty, otherwise lookup will fail.
7841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
7843 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
7846 configure: rename package from gst-rtsp to gst-rtsp-server
7847 To match git module name and avoid confusion with the
7848 rtsp lib in gst-plugins-base and rtsp plugin in -good.
7850 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
7853 configure: bump core/base/good requirement to 1.2.0
7854 Bump to released stable version and make implicit
7855 requirements explicit.
7857 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
7862 Fix broken gettext setup which is not used anyway
7864 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
7867 Automatic update of common submodule
7868 From dbedaa0 to d48bed3
7870 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
7872 * gst/rtsp-server/rtsp-client.c:
7873 * gst/rtsp-server/rtsp-media.c:
7874 * gst/rtsp-server/rtsp-media.h:
7875 media: add setup_sdp vmethod
7876 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
7877 gst_rtsp_media_setup_sdp.
7878 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
7880 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
7882 * gst/rtsp-server/rtsp-stream.c:
7883 rtsp-stream: Check return value of sscanf
7884 streamid is only valid if sscanf matched something.
7886 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
7888 * gst/rtsp-server/rtsp-client.c:
7889 rtsp-client: Fix iteration
7890 Wouldn't even enter the code block otherwise (i++ was used as the check
7891 and not the postfix).
7893 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
7895 * gst/rtsp-server/rtsp-client.c:
7896 * gst/rtsp-server/rtsp-client.h:
7897 client: add vmethod to configure media and streams
7898 Implement a vmethod that can be used to configure the media and the
7899 streams based on the current context. Handle the blocksize handling in
7900 the default handler.
7901 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
7903 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7906 Make git ignore more unit test binaries
7908 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7910 * gst/rtsp-server/rtsp-address-pool.h:
7911 * gst/rtsp-server/rtsp-auth.h:
7912 * gst/rtsp-server/rtsp-client.h:
7913 * gst/rtsp-server/rtsp-context.h:
7914 * gst/rtsp-server/rtsp-media-factory-uri.h:
7915 * gst/rtsp-server/rtsp-media-factory.h:
7916 * gst/rtsp-server/rtsp-media.h:
7917 * gst/rtsp-server/rtsp-mount-points.h:
7918 * gst/rtsp-server/rtsp-server.h:
7919 * gst/rtsp-server/rtsp-session-media.h:
7920 * gst/rtsp-server/rtsp-session-pool.h:
7921 * gst/rtsp-server/rtsp-session.h:
7922 * gst/rtsp-server/rtsp-stream-transport.h:
7923 * gst/rtsp-server/rtsp-stream.h:
7924 * gst/rtsp-server/rtsp-thread-pool.h:
7925 * gst/rtsp-server/rtsp-token.h:
7926 rtsp-server: add padding to many public structures
7927 Not mini objects though, since they are not subclassable
7928 anyway, nor kept on the stack or inlined in a structure.
7930 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7932 media: add new create_rtpbin vmethod
7933 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
7934 https://bugzilla.gnome.org/show_bug.cgi?id=719734
7936 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
7938 * tests/check/gst/media.c:
7939 tests: fix memory leak, free test's thread pool
7940 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
7942 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
7944 * gst/rtsp-server/rtsp-stream-transport.c:
7945 stream-transport: free url in finalize
7947 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
7949 * gst/rtsp-server/rtsp-media.c:
7950 media: also do state change in suspended state
7952 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
7954 * gst/rtsp-server/rtsp-client.c:
7955 * gst/rtsp-server/rtsp-media.c:
7956 media: also handle prepare and range in suspended state
7957 When we are suspended, we are already prepared.
7958 We can get the range in the suspended state.
7960 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
7962 * tests/check/Makefile.am:
7963 * tests/check/gst/sessionmedia.c:
7964 check: add test for uri in setup
7965 Added unit tests for the new functionality in GstRTSPStreamTransport.
7966 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7968 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
7970 * gst/rtsp-server/rtsp-client.c:
7971 client: store setup uri and use in PLAY response
7972 Store the uri used when doing the setup and use that in the PLAY
7974 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7976 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
7978 * gst/rtsp-server/rtsp-stream-transport.c:
7979 * gst/rtsp-server/rtsp-stream-transport.h:
7980 stream-transport: add method to get/set url
7982 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
7984 * gst/rtsp-server/rtsp-client.c:
7985 client: suspend after SDP and unsuspend before PLAYING
7986 Based on patches by Ognyan Tonchev <ognyan@axis.com>
7987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
7989 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
7991 * gst/rtsp-server/rtsp-media-factory.c:
7992 * gst/rtsp-server/rtsp-media-factory.h:
7993 * gst/rtsp-server/rtsp-media.c:
7994 * gst/rtsp-server/rtsp-media.h:
7995 * gst/rtsp-server/rtsp-session-media.c:
7996 * gst/rtsp-server/rtsp-session.c:
7997 * tests/check/gst/media.c:
7998 * tests/check/gst/mediafactory.c:
7999 media: add suspend modes
8000 Add support for different suspend modes. The stream is suspended right after
8001 producing the SDP and after PAUSE. Different suspend modes are available that
8002 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8003 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8004 state and RESET will bring the pipeline to the NULL state.
8005 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8006 this means that the pipeline needs to be prerolled again.
8007 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8008 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8010 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8012 * gst/rtsp-server/rtsp-media.c:
8013 media: start live streams in blocked state
8014 Start live streams in the blocked state and make them preroll using the
8015 messages. This ensure that no data is played by the sink until we explicitly
8016 unblock the stream right before going to PLAYING.
8017 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8019 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8021 * gst/rtsp-server/rtsp-media.c:
8022 media: refactor starting and waiting for preroll
8023 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8024 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8026 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8028 * gst/rtsp-server/rtsp-stream.c:
8029 * gst/rtsp-server/rtsp-stream.h:
8030 stream: add API to block streams
8031 Add an API to block on the streams and make it post a message.
8032 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8033 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8035 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8037 * docs/libs/Makefile.am:
8038 docs: Specify the override file
8039 Even if it's empty (for now) it avoids make distcheck complaining
8041 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8043 * gst/rtsp-server/rtsp-media.c:
8044 media: move default implementations to where they are used
8046 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8048 * gst/rtsp-server/rtsp-media.c:
8049 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8050 We need to take the state_lock when calling this method.
8052 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8054 * gst/rtsp-server/rtsp-media.c:
8055 media: handle add-added on non-bins too
8056 Handle dynamic payloaders that are not bins, as used in the unit-test.
8058 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8060 * gst/rtsp-server/rtsp-media-factory.c:
8061 * gst/rtsp-server/rtsp-media-factory.h:
8062 * gst/rtsp-server/rtsp-media.c:
8063 rtsp-media/-factory: Fix request pad name comments
8064 These must be escaped for gtk-doc to parse the comments without warnings.
8066 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8068 rtsp-media: remove transports if media is in error status
8069 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8070 trying to change to GST_STATE_NULL and media is in error status, we
8071 remove all transports.
8072 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8074 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8076 * gst/rtsp-server/rtsp-media.c:
8077 rtsp-media: use element metadata to find payloader
8078 Use the element metadata to find the payloader instead of checking
8080 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8082 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8084 rtsp-stream: add getter for payload type
8085 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8086 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8087 element and create the stream with this one instead of the dynpay%d
8089 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8091 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8093 * gst/rtsp-server/rtsp-client.c:
8094 * gst/rtsp-server/rtsp-context.h:
8095 * gst/rtsp-server/rtsp-media.c:
8096 * gst/rtsp-server/rtsp-mount-points.c:
8097 * gst/rtsp-server/rtsp-server.c:
8098 * gst/rtsp-server/rtsp-token.c:
8099 rtsp-*: Refer to NULL as a constant in comments
8101 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8103 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8105 rtsp-*: Fix type name typos in comments
8106 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8107 * rtsp-auth: Refer to part of constant name as text
8108 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8109 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8110 * rtsp-stream: Fix typo when refering to GstBin
8111 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8113 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8116 * docs/libs/gst-rtsp-server-docs.sgml:
8117 * docs/libs/gst-rtsp-server-sections.txt:
8118 docs: Improve documentation
8119 * Include annotation-glossary to quiet gtk-doc
8120 * Rename remaining ClientState -> Context
8121 * Rename object hierarchy file
8122 * Remove stale chapter references
8123 * Add missing function and object references
8124 * Include missing GstRTSPAddressPoolResult
8125 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8127 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8129 * gst/rtsp-server/rtsp-client.c:
8130 * gst/rtsp-server/rtsp-server.c:
8131 * gst/rtsp-server/rtsp-session-pool.c:
8132 * gst/rtsp-server/rtsp-session.c:
8133 * gst/rtsp-server/rtsp-stream.c:
8134 rtsp-server: sprinkle some allow-none annotations for g-i
8136 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8138 * gst/rtsp-server/rtsp-stream.c:
8139 * gst/rtsp-server/rtsp-stream.h:
8140 stream: add method to filter transports
8141 Add a method to safely iterate and collect the stream transports
8142 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8144 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8146 * gst/rtsp-server/rtsp-client.c:
8147 * gst/rtsp-server/rtsp-server.c:
8148 * gst/rtsp-server/rtsp-session-pool.c:
8149 * gst/rtsp-server/rtsp-session.c:
8150 rtsp: allow NULL func in filters
8151 Passing a null function make the filters return a list of
8154 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8156 * gst/rtsp-server/rtsp-address-pool.c:
8157 * tests/check/gst/addresspool.c:
8158 address-pool: fix address increment
8159 Use a guint instead of guint8 to increment the address. It's still not
8160 completely correct because a guint might not be able to hold the complete
8161 address range, but that's an enhacement for later.
8162 Add unit test to test improved behaviour.
8163 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8165 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8167 * gst/rtsp-server/rtsp-client.c:
8168 * tests/check/gst/client.c:
8169 client: allow absolute path in requests
8170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8172 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8174 * gst/rtsp-server/rtsp-client.c:
8175 * gst/rtsp-server/rtsp-client.h:
8176 client: make make_path_from_uri a vmethod
8178 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8180 * docs/libs/gst-rtsp-server-sections.txt:
8181 * gst/rtsp-server/rtsp-stream.c:
8182 * gst/rtsp-server/rtsp-stream.h:
8183 * tests/check/Makefile.am:
8184 * tests/check/gst/stream.c:
8185 stream: Add functions to get rtp and rtcp sockets
8186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8188 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8190 * gst/rtsp-server/rtsp-context.c:
8191 * gst/rtsp-server/rtsp-context.h:
8192 context: defing a GType for the context
8193 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8195 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8197 * gst/rtsp-server/Makefile.am:
8198 * gst/rtsp-server/rtsp-auth.c:
8199 * gst/rtsp-server/rtsp-context.c:
8200 * gst/rtsp-server/rtsp-media.c:
8201 * gst/rtsp-server/rtsp-mount-points.c:
8202 * gst/rtsp-server/rtsp-server.h:
8203 * gst/rtsp-server/rtsp-session-media.c:
8204 * gst/rtsp-server/rtsp-session.c:
8205 * gst/rtsp-server/rtsp-stream.c:
8206 Fixed several GIR warnings
8208 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8210 * gst/rtsp-server/rtsp-auth.c:
8213 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8215 * tests/check/Makefile.am:
8216 * tests/check/gst/token.c:
8217 tests: Add unit tests for token
8218 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8220 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8222 * gst/rtsp-server/rtsp-token.c:
8223 token: Validate args for gst_rtsp_token_is_allowed
8224 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8226 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8228 * gst/rtsp-server/rtsp-token.c:
8229 token: Fix bug when creating empty token
8230 We always want to have a valid GstStructure in the token.
8231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8233 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8235 * gst/rtsp-server/rtsp-thread-pool.c:
8236 thread-pool: avoid race in shutdown
8237 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8238 don't actually stop the mainloop ever. Solve this race by adding an idle source
8239 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8240 if quit was called before we started it.
8242 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8244 * tests/check/Makefile.am:
8245 * tests/check/gst/permissions.c:
8246 tests: Add unit tests for permissions
8247 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8249 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8251 * tests/check/gst/mediafactory.c:
8252 tests: Test mediafactory permissions
8253 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8255 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8257 * gst/rtsp-server/rtsp-permissions.c:
8258 permissions: Fix refcounting when adding/removing roles
8259 Previously a role that was removed was unreffed twice, and when
8260 replacing an existing role the replaced role was freed while still being
8261 referenced. Both bugs are now fixed.
8262 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8264 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8266 * tests/check/gst/media.c:
8267 * tests/check/gst/mediafactory.c:
8268 * tests/check/gst/rtspserver.c:
8269 tests: Check gst_rtsp_url_parse return value
8270 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8272 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8275 Automatic update of common submodule
8276 From 865aa20 to dbedaa0
8278 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8280 * gst/rtsp-server/rtsp-server.c:
8281 rtsp-server: Fix socket leak
8282 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8284 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8286 * gst/rtsp-server/rtsp-session-pool.c:
8287 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8288 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8290 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8292 * examples/.gitignore:
8293 * examples/test-video.c:
8294 examples: fix compilation when WITH_AUTH is defined
8295 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8297 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8300 gitignore: Add new test binary
8302 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8304 * tests/check/Makefile.am:
8305 * tests/check/gst/threadpool.c:
8306 thread-pool: Add unit test for the thread pools
8307 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8309 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8311 * gst/rtsp-server/rtsp-thread-pool.c:
8312 thread-pool: Fix thread leak when reusing threads
8313 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8315 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8317 * gst/rtsp-server/rtsp-server.c:
8318 * tests/check/gst/rtspserver.c:
8319 tests: fixed racy behavior in rtspserver tests
8320 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8322 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8324 * tests/check/gst/addresspool.c:
8325 tests: Improve address pool unit tests
8326 Add a range with mixed IPV4 and IPV6 addresses to pool.
8327 Get an IPV4 address from an IPV6-only pool.
8328 Get an IPV6 address from an IPV4-only pool.
8329 Reserve a IPV6 address from an IPV4-only pool.
8330 Check for unicast addresses in multicast-only pool.
8331 Check for unicast addresses in uni-/multicast-mixed pool.
8332 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8334 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8336 * gst/rtsp-server/rtsp-client.c:
8337 client: append query string in PAUSE/PLAY/TEARDOWN as well
8339 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8341 * gst/rtsp-server/rtsp-client.c:
8342 client: Add query to control path
8343 If the SETUP url contains a query it must be appended to the control
8344 path so that it matches any already created stream in the media. The
8345 query will also be appended to the session media path.
8347 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8349 * gst/rtsp-server/rtsp-media.c:
8350 rtsp-media: remove old line
8352 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8354 * gst/rtsp-server/rtsp-stream.c:
8355 stream: Correct control comparison
8356 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8358 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8360 * gst/rtsp-server/rtsp-media.c:
8361 media: Check dynamically if the pipeline supports seeking
8362 We should not depend on whether or not the pipeline state change
8363 returned NO_PREROLL or not. A media could dynamically change its
8364 element and switch from seekable to non seekable so it's best to test
8365 the seekable nature of the pipeline dynamically when we try to do a seek.
8367 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8369 * gst/rtsp-server/rtsp-media.c:
8370 media: Return FALSE if seeking is not supported
8372 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8374 * gst/rtsp-server/rtsp-media.c:
8375 rtsp-media: don't seek accurate by default
8376 Accurate seeking is perhaps a little overkill in the most common situation and
8377 causes some formats (mp3) over slow media to seek extremely slowly.
8379 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8381 * tests/check/gst/rtspserver.c:
8382 tests: fix unit test
8383 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8385 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8387 * gst/rtsp-server/rtsp-client.c:
8388 client: Reply 400 if media cannot be constructed
8389 Reply 400 Bad Request instead of 503 Service Unavailable if media
8390 cannot be constructed in SETUP.
8391 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8393 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8395 * gst/rtsp-server/rtsp-client.c:
8396 client: Send setup reply once only
8397 If find_media() failed in handle_setup_request() two replies was sent.
8398 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8400 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8403 Automatic update of common submodule
8404 From 6b03ba7 to 865aa20
8406 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8408 * gst/rtsp-server/rtsp-server.c:
8409 server: Emit client-connected signal earlier
8410 Emit client-connected before the client ref is given to a GSource,
8411 otherwise client-connected can be emitted after the client object has
8414 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8416 * gst/rtsp-server/rtsp-address-pool.c:
8417 * gst/rtsp-server/rtsp-address-pool.h:
8418 * gst/rtsp-server/rtsp-stream.c:
8419 * tests/check/gst/addresspool.c:
8420 addresspool: return reason of failure
8421 Let gst_rtsp_address_pool_reserve_address() return the reason why
8422 the address could not be reserved.
8423 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8425 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8428 autogen.sh: Sync behaviour with other GStreamer modules
8429 Allows building from outside of tree amongst other things
8431 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8434 Automatic update of common submodule
8435 From b613661 to 6b03ba7
8437 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8440 Automatic update of common submodule
8441 From 74a6857 to b613661
8443 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8446 Automatic update of common submodule
8447 From 01a7a46 to 74a6857
8449 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8451 * gst/rtsp-server/rtsp-client.c:
8452 client: Do not read beyond end of path string
8453 If the setup was done without a control url, make sure we don't try to read the
8454 non-existing control string and crash.
8456 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8458 * gst/rtsp-server/rtsp-client.c:
8459 client: Fix RTPInfo header
8460 Refactor the method to make the content_base.
8461 Use the content-base and the control url to construct the RTPInfo
8464 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8466 * gst/rtsp-server/rtsp-client.c:
8467 client: map url to path only in describe
8468 Only map the request url to a path in the DESCRIBE method. The SDP then
8469 contains the base and control urls that should be used to SETUP/PAUSE/
8470 PLAY/TEARDOWN the media.
8472 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8474 * gst/rtsp-server/rtsp-client.c:
8475 Revert "client: map URL to path in requests"
8476 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8477 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8478 contains the base and control urls which are used in the SETUP, PLAY,
8479 PAUSE and TEARDOWN requests.
8481 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8483 * gst/rtsp-server/rtsp-client.c:
8484 client: map URL to path in requests
8486 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8488 * gst/rtsp-server/rtsp-client.c:
8489 * gst/rtsp-server/rtsp-mount-points.c:
8490 * gst/rtsp-server/rtsp-mount-points.h:
8491 mount-points: make vmethod to make path from uri
8492 Make a vmethod to transform an url into a path. The path is then used to lookup
8493 the factory. This makes it possible to also use other bits of the url, such as
8494 the query parameters, to locate the factory.
8496 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8498 * gst/rtsp-server/rtsp-thread-pool.c:
8499 * gst/rtsp-server/rtsp-thread-pool.h:
8500 thread-pool: Add cleanup to wait for the threadpool to finish
8501 Also fix race condition if two threads are asking for the first
8502 thread from the thread pool at once. This would case two internal
8503 GThreadPools to be created.
8504 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8506 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8508 * gst/rtsp-server/rtsp-client.c:
8509 * tests/check/gst/client.c:
8510 client: free threadpool
8511 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8513 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8515 * tests/check/gst/mountpoints.c:
8516 mountpoints tests: unref matched factories
8517 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8519 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8521 * tests/check/gst/media.c:
8522 media tests: unref thread pool and caps
8523 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8525 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8527 * gst/rtsp-server/rtsp-auth.c:
8528 * gst/rtsp-server/rtsp-media-factory.c:
8529 * gst/rtsp-server/rtsp-media.c:
8530 auth, media, media-factory: unref permissions
8531 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8533 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8535 * examples/Makefile.am:
8536 Makefile: add rule for appsrc example
8538 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * examples/test-appsrc.c:
8541 tests: add appsrc example
8542 Add an example on how to use appsrc to feed the server pipeline with data.
8544 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8546 * gst/rtsp-server/rtsp-client.c:
8547 rtsp-client: remove query part from content-base string
8548 Make sure that after the control url has been resolved, it's
8549 not a part of the query-string.
8550 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8552 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8554 * gst/rtsp-server/rtsp-client.c:
8555 client: don't check url in response
8556 There is no url or method in the response to check
8558 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8560 * gst/rtsp-server/rtsp-client.c:
8561 * gst/rtsp-server/rtsp-client.h:
8562 Add handle-response signal for when we receive a GET_PARAMETER response
8564 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8566 * gst/rtsp-server/rtsp-server.c:
8567 Fix gst_rtsp_server_client_filter, using wrong variable type
8569 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8571 * gst/rtsp-server/rtsp-media-factory-uri.c:
8572 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8573 For AAC we need to check for framed=true instead of parsed=true.
8574 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8576 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * gst/rtsp-server/rtsp-stream.c:
8579 stream: optimize pipeline for protocols
8580 When TCP is not an allowed protocol for the stream, avoid creating the
8581 appsrc/appsink/queue and tee elements.
8583 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8585 * gst/rtsp-server/rtsp-media.c:
8586 media: set protocols on streams
8588 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8590 * gst/rtsp-server/rtsp-client.c:
8591 client: use protocols supported by stream
8593 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8595 * gst/rtsp-server/rtsp-media-factory.c:
8596 * gst/rtsp-server/rtsp-media.c:
8597 * gst/rtsp-server/rtsp-stream.c:
8598 media-factory: allow all protocols
8600 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8602 * gst/rtsp-server/rtsp-media.c:
8603 media: configure protocols in new streams
8605 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8607 * gst/rtsp-server/rtsp-stream.c:
8608 * gst/rtsp-server/rtsp-stream.h:
8609 stream: add protocols property
8611 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8613 * gst/rtsp-server/rtsp-media.c:
8614 rtsp-media: send state in "new-state" signal
8615 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8617 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8620 build: add subdir-objects to AM_INIT_AUTOMAKE
8621 Fixes warnings with automake 1.14
8622 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8624 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * docs/libs/gst-rtsp-server-sections.txt:
8627 * gst/rtsp-server/rtsp-client.c:
8628 * gst/rtsp-server/rtsp-server.c:
8629 * gst/rtsp-server/rtsp-server.h:
8630 server: add method to iterate clients of server
8632 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8634 * gst/rtsp-server/rtsp-media.c:
8635 * gst/rtsp-server/rtsp-media.h:
8636 Add vmethod for rtsp-media subclass to access rtpbin
8638 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-client.h:
8641 small documentation fix
8643 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8645 * gst/rtsp-server/rtsp-client.c:
8646 Do not take range header if range is invalid
8648 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8650 * docs/libs/gst-rtsp-server-sections.txt:
8651 * gst/rtsp-server/rtsp-media.c:
8652 media: add docs for new method
8654 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8656 * gst/rtsp-server/rtsp-media.c:
8657 * gst/rtsp-server/rtsp-media.h:
8658 Add API to rtsp-media set the pipeline's state
8660 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-media.c:
8663 Update current position/duration when gst_rtsp_media_get_range_string is called
8665 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8667 * examples/test-cgroups.c:
8668 tests: add some more docs
8670 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8672 * examples/test-cgroups.c:
8673 * gst/rtsp-server/Makefile.am:
8674 * gst/rtsp-server/rtsp-auth.c:
8675 * gst/rtsp-server/rtsp-auth.h:
8676 * gst/rtsp-server/rtsp-client.c:
8677 * gst/rtsp-server/rtsp-client.h:
8678 * gst/rtsp-server/rtsp-context.c:
8679 * gst/rtsp-server/rtsp-context.h:
8680 * gst/rtsp-server/rtsp-params.c:
8681 * gst/rtsp-server/rtsp-params.h:
8682 * gst/rtsp-server/rtsp-server.c:
8683 * gst/rtsp-server/rtsp-thread-pool.c:
8684 * gst/rtsp-server/rtsp-thread-pool.h:
8685 * tests/check/gst/client.c:
8686 ClientState -> Context
8687 Rename the clientstate to context and put the code in a separate file.
8689 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8691 * examples/test-auth.c:
8692 * gst/rtsp-server/rtsp-auth.c:
8693 * gst/rtsp-server/rtsp-auth.h:
8694 auth: add support for default token
8695 The default token is used when the user is not authenticated and can be used to
8696 give minimal permissions.
8698 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8700 * examples/test-auth.c:
8701 * gst/rtsp-server/rtsp-auth.c:
8702 auth: use defines when possible
8704 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8706 * gst/rtsp-server/rtsp-address-pool.c:
8707 address-pool: improve docs
8709 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8711 * gst/rtsp-server/rtsp-permissions.c:
8712 permissions: add the role to the copy
8714 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
8716 * gst/rtsp-server/rtsp-permissions.c:
8717 permissions: Also copy the roles
8719 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
8721 * gst/rtsp-server/rtsp-permissions.c:
8722 permissions: Make it build
8724 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8726 * gst/rtsp-server/rtsp-address-pool.h:
8729 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8731 * docs/libs/gst-rtsp-server-sections.txt:
8732 * gst/rtsp-server/rtsp-auth.c:
8733 * gst/rtsp-server/rtsp-auth.h:
8734 * gst/rtsp-server/rtsp-media.c:
8735 * gst/rtsp-server/rtsp-session-media.c:
8736 * gst/rtsp-server/rtsp-stream-transport.c:
8737 * gst/rtsp-server/rtsp-stream-transport.h:
8738 * gst/rtsp-server/rtsp-stream.c:
8739 * tests/check/gst/client.c:
8742 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8744 * docs/libs/gst-rtsp-server-sections.txt:
8745 * gst/rtsp-server/rtsp-address-pool.c:
8746 * gst/rtsp-server/rtsp-address-pool.h:
8747 * tests/check/gst/addresspool.c:
8748 * tests/check/gst/rtspserver.c:
8749 address-pool: cleanups
8750 Remove redundant method, improve docs.
8752 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8754 * docs/libs/gst-rtsp-server-sections.txt:
8755 * gst/rtsp-server/rtsp-auth.h:
8756 * gst/rtsp-server/rtsp-permissions.c:
8757 * gst/rtsp-server/rtsp-permissions.h:
8758 * gst/rtsp-server/rtsp-token.c:
8761 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8763 * gst/rtsp-server/rtsp-permissions.c:
8764 permissions: implement _remove_role
8766 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8768 * gst/rtsp-server/rtsp-permissions.c:
8769 permissions: update docs
8771 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8773 * tests/check/gst/client.c:
8774 tests: simplify tests
8775 Client settings are now disabled by default so we don't need an auth
8776 module to disable them.
8778 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8780 * gst/rtsp-server/rtsp-auth.c:
8781 auth: add default authorizations
8782 When no auth module is specified, use our table of defaults to look up the
8783 default value of the check instead of always allowing everything. This was
8784 we can disallow client settings by default.
8786 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8789 README: update readme
8791 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8793 * gst/rtsp-server/rtsp-thread-pool.c:
8794 * gst/rtsp-server/rtsp-thread-pool.h:
8795 thread-pool: add more docs
8797 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8799 * gst/rtsp-server/rtsp-thread-pool.c:
8800 * gst/rtsp-server/rtsp-thread-pool.h:
8801 thread-pool: fix race in thread reuse
8802 If we try to reuse a thread right after we made it stop, we end up using a
8803 stopped thread. Catch this case and only reuse threads that are not stopping.
8805 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8807 * gst/rtsp-server/rtsp-server.c:
8808 server: add small debug
8810 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8812 * tests/check/gst/client.c:
8814 Add some permissions to media so we can use the auth and enable
8817 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8819 * gst/rtsp-server/rtsp-client.c:
8820 client: support pushed context in handle_request
8821 If we already have a pushed state, reuse it and add our own things. This makes
8822 it easier to write tests.
8824 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8826 * gst/rtsp-server/rtsp-auth.c:
8827 auth: don't auth on methods
8828 Don't authorize on methods anymore but on the resources that we
8829 try to access, this is more flexible.
8830 Move the authorization checks to where they are needed and let the
8831 check return the response on error.
8833 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8835 * gst/rtsp-server/rtsp-mount-points.c:
8836 mount-points: add some debug
8838 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8840 * tests/check/gst/client.c:
8841 tests: almost fix test
8843 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8845 * gst/rtsp-server/rtsp-auth.c:
8846 * gst/rtsp-server/rtsp-auth.h:
8847 * gst/rtsp-server/rtsp-client.c:
8848 * gst/rtsp-server/rtsp-client.h:
8849 * gst/rtsp-server/rtsp-server.c:
8850 * gst/rtsp-server/rtsp-server.h:
8851 auth: let the auth module check client_settings
8852 Let the auth module decide if client settings are allowed for the
8855 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8857 * gst/rtsp-server/rtsp-token.c:
8858 * gst/rtsp-server/rtsp-token.h:
8859 token: add method to check boolean permission
8861 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8863 * examples/test-auth.c:
8864 * examples/test-cgroups.c:
8865 * gst/rtsp-server/rtsp-token.c:
8866 * gst/rtsp-server/rtsp-token.h:
8867 token: simplify token constructor
8868 Use variable arguments to make easier API.
8870 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8872 * examples/test-auth.c:
8873 * examples/test-cgroups.c:
8874 * gst/rtsp-server/rtsp-media-factory.c:
8875 * gst/rtsp-server/rtsp-media-factory.h:
8876 media-factory: add convenience API for factory
8878 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * examples/test-auth.c:
8881 * examples/test-cgroups.c:
8882 * gst/rtsp-server/rtsp-permissions.c:
8883 * gst/rtsp-server/rtsp-permissions.h:
8884 permissions: simplify API a little
8885 Avoid passing GstStructure in the add_role method, use varargs instead
8886 to construct the structure behind the scenes. We can then also use the
8887 structure name as the role and simplify some more logic.
8889 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8891 * gst/rtsp-server/rtsp-auth.c:
8894 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8896 * gst/rtsp-server/rtsp-auth.c:
8897 * gst/rtsp-server/rtsp-auth.h:
8898 * gst/rtsp-server/rtsp-client.c:
8899 auth: handle unauthorized response
8900 Move handling of the unauthorized response to the auth module, it can add
8901 the appropriate headers to request authorization for the required method
8902 much better than the client.
8904 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8906 * gst/rtsp-server/rtsp-client.c:
8907 * gst/rtsp-server/rtsp-client.h:
8908 client: allow for sending any message, not only requests
8909 Change the _send_request() method to _send_message() so that we
8910 can both send requests and replies.
8912 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8914 * docs/libs/gst-rtsp-server-sections.txt:
8915 * gst/rtsp-server/rtsp-server.h:
8918 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8920 * examples/test-video.c:
8921 * gst/rtsp-server/rtsp-auth.c:
8922 * gst/rtsp-server/rtsp-auth.h:
8923 * gst/rtsp-server/rtsp-server.c:
8924 * gst/rtsp-server/rtsp-server.h:
8925 auth: move TLS handling to auth module
8926 Remove the TLS settings on the server and move it to the auth module because
8927 that is where security related bits go.
8929 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8931 * gst/rtsp-server/rtsp-client.c:
8932 * gst/rtsp-server/rtsp-client.h:
8933 client: add state push/pop
8935 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8937 * gst/rtsp-server/rtsp-client.c:
8938 * gst/rtsp-server/rtsp-client.h:
8939 client: add connection to state
8941 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8943 * gst/rtsp-server/rtsp-mount-points.c:
8944 mount-points: fix debug
8946 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8948 * tests/check/gst/media.c:
8949 tests: fix media test
8951 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8953 * gst/rtsp-server/rtsp-thread-pool.c:
8954 thread-pool: we don't require a state
8956 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8958 * gst/rtsp-server/rtsp-server.c:
8959 server: let context ref the server
8960 So that we don't risk losing the server object early anc crash.
8962 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8964 * tests/check/gst/client.c:
8965 tests: fix client test
8967 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * docs/libs/gst-rtsp-server-docs.sgml:
8971 * docs/libs/gst-rtsp-server-sections.txt:
8972 * gst/rtsp-server/rtsp-address-pool.c:
8973 * gst/rtsp-server/rtsp-auth.c:
8974 * gst/rtsp-server/rtsp-client.c:
8975 * gst/rtsp-server/rtsp-client.h:
8976 * gst/rtsp-server/rtsp-media-factory-uri.c:
8977 * gst/rtsp-server/rtsp-media-factory.c:
8978 * gst/rtsp-server/rtsp-media-factory.h:
8979 * gst/rtsp-server/rtsp-media.c:
8980 * gst/rtsp-server/rtsp-mount-points.c:
8981 * gst/rtsp-server/rtsp-params.c:
8982 * gst/rtsp-server/rtsp-permissions.c:
8983 * gst/rtsp-server/rtsp-sdp.c:
8984 * gst/rtsp-server/rtsp-server.c:
8985 * gst/rtsp-server/rtsp-server.h:
8986 * gst/rtsp-server/rtsp-session-media.c:
8987 * gst/rtsp-server/rtsp-session-pool.c:
8988 * gst/rtsp-server/rtsp-session.c:
8989 * gst/rtsp-server/rtsp-stream-transport.c:
8990 * gst/rtsp-server/rtsp-stream.c:
8991 * gst/rtsp-server/rtsp-thread-pool.c:
8992 * gst/rtsp-server/rtsp-token.c:
8995 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8997 * gst/rtsp-server/rtsp-session-pool.c:
8998 * gst/rtsp-server/rtsp-session-pool.h:
8999 session-pool: make vmethod to create a session
9000 Make a vmethod to create a sessions so that subclasses can create
9001 custom session objects
9003 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9005 * gst/rtsp-server/rtsp-auth.c:
9006 * gst/rtsp-server/rtsp-media-factory.h:
9007 * gst/rtsp-server/rtsp-media.h:
9008 * gst/rtsp-server/rtsp-mount-points.h:
9009 * gst/rtsp-server/rtsp-session-pool.h:
9010 * gst/rtsp-server/rtsp-stream.h:
9013 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9015 * docs/libs/gst-rtsp-server-docs.sgml:
9016 * docs/libs/gst-rtsp-server-sections.txt:
9017 * gst/rtsp-server/rtsp-address-pool.c:
9018 * gst/rtsp-server/rtsp-address-pool.h:
9019 * gst/rtsp-server/rtsp-auth.c:
9020 * gst/rtsp-server/rtsp-client.h:
9021 * gst/rtsp-server/rtsp-media-factory.h:
9022 * gst/rtsp-server/rtsp-media.c:
9023 * gst/rtsp-server/rtsp-media.h:
9024 * gst/rtsp-server/rtsp-permissions.c:
9025 * gst/rtsp-server/rtsp-permissions.h:
9026 * gst/rtsp-server/rtsp-server.h:
9027 * gst/rtsp-server/rtsp-session-media.c:
9028 * gst/rtsp-server/rtsp-session-media.h:
9029 * gst/rtsp-server/rtsp-session-pool.h:
9030 * gst/rtsp-server/rtsp-session.h:
9031 * gst/rtsp-server/rtsp-stream-transport.h:
9032 * gst/rtsp-server/rtsp-stream.c:
9033 * gst/rtsp-server/rtsp-thread-pool.h:
9036 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9039 * examples/Makefile.am:
9040 configure: compile cgroup example conditionally
9041 Only compile the cgroup example when we have libcgroup
9043 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * examples/Makefile.am:
9047 * examples/test-cgroups.c:
9048 examples: add cgroups example
9050 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9052 * tests/check/gst/rtspserver.c:
9053 tests: fix compilation
9055 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9057 * gst/rtsp-server/rtsp-thread-pool.c:
9058 thread-pool: fix vmethod invocation
9060 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9062 * gst/rtsp-server/rtsp-thread-pool.c:
9063 * gst/rtsp-server/rtsp-thread-pool.h:
9064 thread-pool: store thread type in thread
9066 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9068 * gst/rtsp-server/rtsp-client.c:
9069 client: pass thread from pool to media _prepare
9070 Get a thread from the configured threadpool and pass it to the prepare method of
9073 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9075 * gst/rtsp-server/rtsp-media.c:
9076 * gst/rtsp-server/rtsp-media.h:
9077 media: Accept a thread in _prepare
9078 Remove out own threadpool handling and use the provided thread and
9079 maincontext for the bus messages and the state changes.
9081 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9083 * gst/rtsp-server/rtsp-server.c:
9084 server: configure client thread pool
9086 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9088 * gst/rtsp-server/rtsp-client.c:
9089 * gst/rtsp-server/rtsp-client.h:
9090 client: add method to configure thread pool
9092 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9094 * gst/rtsp-server/rtsp-client.h:
9095 * gst/rtsp-server/rtsp-server.c:
9096 * gst/rtsp-server/rtsp-server.h:
9097 server: use thread pool
9098 Use the thread pool instead of doing our own thing.
9100 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9102 * gst/rtsp-server/Makefile.am:
9103 * gst/rtsp-server/rtsp-thread-pool.c:
9104 * gst/rtsp-server/rtsp-thread-pool.h:
9105 thread-pool: add object to manage threads
9106 Add an object to manage the client and media threads.
9108 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9110 * gst/rtsp-server/rtsp-auth.c:
9111 auth: debug authorization check
9113 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9115 * gst/rtsp-server/rtsp-media.c:
9116 media: start media pipeline in context
9117 Start the media pipeline in the provided context (or our default one
9118 when NULL). This makes sure that we run the bus thread in this context and that
9119 all media threads are children of this context.
9121 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9123 * gst/rtsp-server/rtsp-media-factory.c:
9124 factory: pass permissions to media by default
9126 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * examples/test-auth.c:
9129 test: add permissions to auth test
9130 Ass some permissions to the media factory in the test.
9132 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9134 * gst/rtsp-server/rtsp-auth.c:
9135 * gst/rtsp-server/rtsp-auth.h:
9136 * gst/rtsp-server/rtsp-client.c:
9137 auth: simplify auth checks
9138 Remove client from methods, it's now in the state
9139 Perform the check specified by the string, use the information from the
9140 thread local context.
9142 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9144 * gst/rtsp-server/rtsp-client.c:
9145 * gst/rtsp-server/rtsp-client.h:
9146 client: add state to current thread
9147 Add the client to the ClientState object.
9148 Place the ClientState on the current thread.
9150 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9152 * gst/rtsp-server/rtsp-media-factory.c:
9153 * gst/rtsp-server/rtsp-media-factory.h:
9154 * gst/rtsp-server/rtsp-media.c:
9155 * gst/rtsp-server/rtsp-media.h:
9156 media: make it possible to set permissions
9157 Make it possible to set permissions on media and media factory objects
9159 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9161 * gst/rtsp-server/Makefile.am:
9162 * gst/rtsp-server/rtsp-permissions.c:
9163 * gst/rtsp-server/rtsp-permissions.h:
9164 permissions: add permissions object
9165 Add a mini object to store permissions based on a role.
9167 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9169 * examples/test-auth.c:
9170 * gst/rtsp-server/rtsp-auth.c:
9171 * gst/rtsp-server/rtsp-auth.h:
9172 * gst/rtsp-server/rtsp-client.c:
9173 auth: add auth checks
9174 Add an enum with auth checks and implement the checks in the auth object.
9175 Perform the checks from the client.
9177 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9179 * examples/test-auth.c:
9180 * gst/rtsp-server/rtsp-auth.c:
9181 * gst/rtsp-server/rtsp-auth.h:
9182 * gst/rtsp-server/rtsp-client.h:
9183 auth: use the token after authentication
9184 After we authenticated a user, keep the Token around in the state.
9186 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9188 * gst/rtsp-server/rtsp-client.c:
9189 * gst/rtsp-server/rtsp-media.c:
9190 * gst/rtsp-server/rtsp-media.h:
9191 * tests/check/gst/media.c:
9192 media: add optional context for bus messages
9193 Add an optional mainloop to _prepare that will handle the bus messages instead
9194 of always using the shared mainloop.
9196 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9198 * gst/rtsp-server/Makefile.am:
9199 * gst/rtsp-server/rtsp-token.c:
9200 * gst/rtsp-server/rtsp-token.h:
9201 token: add authorization token
9202 Add a simply miniobject that contains the authorizations. The object contains a
9203 GstStructure that hold all authorization fields. When a user is authenticated,
9204 the auth module will create a Token for the user. The token is then used to
9205 check what operations the user is allowed to do and various other configuration
9208 2013-07-05 12:08:36 +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.c:
9214 * gst/rtsp-server/rtsp-client.h:
9215 * gst/rtsp-server/rtsp-media-factory.c:
9216 * gst/rtsp-server/rtsp-media-factory.h:
9217 * gst/rtsp-server/rtsp-media.c:
9218 * gst/rtsp-server/rtsp-media.h:
9219 auth: remove auth from media and factory
9220 Remove the auth object from media and factory. We want to have the RTSPClient
9221 authenticate and authorize resources, there is no need to place another auth
9222 manager on the media/factory.
9224 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9226 * examples/test-auth.c:
9227 * gst/rtsp-server/rtsp-auth.c:
9228 * gst/rtsp-server/rtsp-auth.h:
9229 * gst/rtsp-server/rtsp-client.h:
9230 auth: add support for multiple basic auth tokens
9231 Make it possible to add multiple basic authorisation tokens to one authorization
9232 object. Associate with each token an authorization group that will define what
9233 capabilities are allowed.
9235 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9237 * gst/rtsp-server/rtsp-client.c:
9238 client: error out on non-aggregate control
9239 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9241 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9243 * gst/rtsp-server/rtsp-client.c:
9244 client: rework setup request a little
9245 Cache the media in DESCRIBE based on the longest matching path with the uri
9246 that we can find in the mount points.
9247 Rework the setup request a little to get the media from the session or from
9248 the longest matching path, this way we can derive the control string as
9249 everything after the path instead of hardcoding it.
9250 Find the stream based on the control string and only open a session when all
9253 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9255 * gst/rtsp-server/rtsp-media.c:
9256 * gst/rtsp-server/rtsp-media.h:
9257 media: add method to find a stream by control url
9259 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9261 * gst/rtsp-server/rtsp-stream.c:
9262 * gst/rtsp-server/rtsp-stream.h:
9263 stream: add method to check control url of stream
9265 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9267 * gst/rtsp-server/rtsp-client.c:
9268 * gst/rtsp-server/rtsp-session-media.c:
9269 * gst/rtsp-server/rtsp-session-media.h:
9270 * gst/rtsp-server/rtsp-session.c:
9271 * gst/rtsp-server/rtsp-session.h:
9272 session: use path matching for session media
9273 Use a path string instead of a uri to lookup session media in the sessions. Also
9274 use path matching to find the largest possible path that matches.
9276 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9278 * gst/rtsp-server/rtsp-client.c:
9279 * gst/rtsp-server/rtsp-mount-points.c:
9280 * gst/rtsp-server/rtsp-mount-points.h:
9281 * tests/check/gst/mountpoints.c:
9282 mount-points: remove useless vmethod
9283 Making lookups in the mount points should not be done with a URL, if there is a
9284 mapping to be done from URL to mount points, we'll need to do it somewhere
9287 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9289 * gst/rtsp-server/rtsp-mount-points.c:
9290 * gst/rtsp-server/rtsp-mount-points.h:
9291 * tests/check/gst/mountpoints.c:
9292 mount-points: improve mount point searching
9293 Use a GSequence to keep track of the mount points.
9294 Match a URL to the longest matching registered mount point. This should be the
9295 URL to perform aggreagate control and the remainder is the stream specific
9297 Add some unit tests for this.
9299 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9301 * gst/rtsp-server/Makefile.am:
9302 rtsp-server: Allow building of static library
9304 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9306 * tests/check/gst/mediafactory.c:
9307 tests: fix compilation
9309 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9311 * gst/rtsp-server/rtsp-sdp.c:
9312 sdp: get control string from stream
9313 Use the control string as configured in the stream.
9315 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9317 * gst/rtsp-server/rtsp-stream.c:
9318 * gst/rtsp-server/rtsp-stream.h:
9319 stream: add methods and property to set control string
9321 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9323 * gst/rtsp-server/rtsp-client.c:
9325 Rename variables for clarity
9326 Keep media in state when we can
9328 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9330 * gst/rtsp-server/rtsp-client.c:
9331 * gst/rtsp-server/rtsp-stream.c:
9332 * gst/rtsp-server/rtsp-stream.h:
9333 stream: add more support for IPv6
9334 Rename _get_address to _get_multicast_address in GstRTSPStream to
9335 make it clear that this function only deals with multicast.
9336 Make it possible to have both an IPv4 and IPv6 multicast address on
9337 a stream. Give the client an IPv4 or IPv6 address depending on the
9338 address it used to connect to the server.
9339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9341 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9343 * gst/rtsp-server/rtsp-client.c:
9346 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9348 * gst/rtsp-server/rtsp-stream.c:
9349 stream: handle failed port allocation
9350 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9351 can't allocate any family at all. Also keep track of what port families we
9353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9355 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9357 * gst/rtsp-server/rtsp-stream.c:
9358 stream: improve docs
9360 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9362 * gst/rtsp-server/rtsp-stream-transport.c:
9363 stream-transport: remove old if 0 block
9365 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9367 * tests/check/gst/client.c:
9369 gst_rtsp_client_get_uri() has been removed
9370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9372 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9374 * gst/rtsp-server/rtsp-client.c:
9375 * gst/rtsp-server/rtsp-client.h:
9376 client: add method to filter managed sessions
9377 Add a method to filter the sessions managed by this client connection.
9378 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9380 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9382 * gst/rtsp-server/rtsp-client.c:
9383 * gst/rtsp-server/rtsp-client.h:
9384 client: remove _get_uri() method
9385 Remove the get_uri() method on the client. A client has no uri, the uri
9386 property is an internal property to manage the last cached media for
9389 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9391 * gst/rtsp-server/rtsp-media-factory.h:
9392 media-factory: fix typo
9394 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9396 * gst/rtsp-server/rtsp-media.c:
9397 rtsp-media: Do not leak the query in default_query_stop
9398 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9400 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9402 * gst/rtsp-server/rtsp-media.c:
9403 media: don't unlock when conversion fails
9404 Don't unlock the state lock when conversion fails because it was not locked.
9406 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9408 * gst/rtsp-server/rtsp-media.c:
9409 * gst/rtsp-server/rtsp-media.h:
9410 Add query_position and query_stop vmethods to rtsp-media
9412 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9414 * gst/rtsp-server/rtsp-media.c:
9415 Fix typo in property install for rtsp-media's time-provider
9417 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9419 * gst/rtsp-server/rtsp-client.c:
9420 * gst/rtsp-server/rtsp-client.h:
9421 client: clean some variables
9422 Clean some variables and add some guards to _send_request()
9424 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9426 * gst/rtsp-server/rtsp-client.c:
9427 * gst/rtsp-server/rtsp-client.h:
9428 Add gst_rtsp_client_send_request API
9429 This makes it possible to send arbitrary messages to a client, such as
9430 SET_PARAMETER or GET_PARAMETER
9432 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9434 * gst/rtsp-server/rtsp-media.c:
9435 * gst/rtsp-server/rtsp-media.h:
9436 media: add _get_element() method
9437 Add method to get the element used when creating the media.
9438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9440 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * gst/rtsp-server/rtsp-media.c:
9445 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9447 * gst/rtsp-server/rtsp-stream.c:
9448 * gst/rtsp-server/rtsp-stream.h:
9449 stream: allow access to the rtp session
9450 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9452 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9454 * gst/rtsp-server/rtsp-stream.c:
9455 * gst/rtsp-server/rtsp-stream.h:
9456 dscp qos support in gst-rtsp-stream
9457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9459 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9461 * tests/check/gst/rtspserver.c:
9463 Actually do what the comment says. Also keep the old code around, not sure what
9464 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9465 it currently doesn't.
9467 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9469 * gst/rtsp-server/rtsp-client.c:
9470 client: also watch newly created session
9471 When we newly created a session, start watching it immediately instead of
9472 on the next request.
9474 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9476 * tests/check/gst/client.c:
9477 tests: add unit test for new-session
9478 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9480 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9482 * gst/rtsp-server/rtsp-client.c:
9483 client: emit new-session when new session is created
9484 Only emit new-session when we created a new session for a client, not when a
9485 client picked up a previous session.
9486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9488 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9490 * gst/rtsp-server/rtsp-client.c:
9491 client: handle asterisk as path in requests
9492 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9494 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9496 * gst/rtsp-server/rtsp-media.c:
9497 media: handle segment query format mismatch
9498 It's possible that the segment query returns with a different format than what
9499 we asked for, handle this case also.
9501 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9503 * gst/rtsp-server/rtsp-media.c:
9504 media: use segment stop in collect_media_stats
9505 Use segment stop instead of duration as range end point.
9506 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9508 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9510 * gst/rtsp-server/rtsp-media.c:
9511 * tests/check/gst/media.c:
9512 rtsp-media: Do not leak the element in take_pipeline
9513 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9515 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9517 * gst/rtsp-server/rtsp-client.c:
9518 * gst/rtsp-server/rtsp-client.h:
9519 rtsp-client: Make configure_client_transport virtual
9520 This patch makes configure_client_transport virtual. The functionality is
9521 needed to handle some weird clients sending multicast transport settings as url
9523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9525 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9527 * gst/rtsp-server/rtsp-client.c:
9528 * gst/rtsp-server/rtsp-client.h:
9529 rtsp-client: Make param_set and param_get virtual
9530 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9532 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9534 * gst/rtsp-server/rtsp-client.c:
9535 * gst/rtsp-server/rtsp-media.c:
9536 * gst/rtsp-server/rtsp-media.h:
9537 media: convert_range replaces get_range_times
9538 get_range_times worked for handling UTC ranges for seeks, but we also
9539 need to convert back from NPT to the requested unit in
9540 get_range_string. convert_range is now used for both.
9541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9543 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9545 * gst/rtsp-server/rtsp-client.c:
9546 * gst/rtsp-server/rtsp-sdp.c:
9547 * gst/rtsp-server/rtsp-sdp.h:
9548 sdp: cleanup sdp info
9549 We don't need to pass the proto, we can more easily check a boolean.
9550 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9552 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9554 * gst/rtsp-server/rtsp-sdp.c:
9555 use 0.0.0.0 or :: for c= line instead of server address
9557 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9559 * gst/rtsp-server/rtsp-client.c:
9560 use local address, not remote, in SDP
9561 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9563 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9566 Automatic update of common submodule
9567 From 098c0d7 to 01a7a46
9569 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9571 * gst/rtsp-server/rtsp-media.c:
9572 * gst/rtsp-server/rtsp-media.h:
9573 media: possibility to override range time conversion
9574 Make it possible to override the conversion from GstRTSPTimeRange to
9575 GstClockTimes, that is done before seeking on the media
9576 pipeline. Overriding can be useful for UTC ranges, where the default
9577 conversion gives nanoseconds since 1900.
9578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9580 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9582 * gst/rtsp-server/rtsp-server.c:
9583 * gst/rtsp-server/rtsp-server.h:
9584 rtsp-server: Expose the use_client_settings API
9585 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9587 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9589 * gst/rtsp-server/rtsp-client.c:
9590 * gst/rtsp-server/rtsp-stream.c:
9591 * gst/rtsp-server/rtsp-stream.h:
9592 rtspstream: handle both ipv4 and ipv6 clients
9593 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9595 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9597 * gst/rtsp-server/rtsp-sdp.c:
9598 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9599 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9600 We already have a way to place extra attributes in the SDP by using a string
9601 property with prefix x- or a- in the caps.
9603 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9605 * gst/rtsp-server/rtsp-sdp.c:
9606 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9607 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9608 We already have a way to place extra attributes in the SDP, just make a string
9609 property in the payloader with a- or x- prefix.
9611 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9613 * gst/rtsp-server/rtsp-sdp.c:
9614 rtsp: place a- and x- properties as attributes
9615 application/x-rtp has properties with a- and x- prefixes that should be
9616 placed as attributes in the SDP for the media instead of being added to the
9619 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9621 * examples/Makefile.am:
9622 * examples/test-video.c:
9623 example: add TLS example
9625 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9627 * gst/rtsp-server/rtsp-server.c:
9628 * gst/rtsp-server/rtsp-server.h:
9629 server: add support for TLS
9630 Add methods to set and get a TLS certificate.
9631 Add vmethod to configure a new connection. By default, configure the TLS
9632 certificate in a new connection if needed.
9634 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9636 * gst/rtsp-server/rtsp-server.c:
9637 * gst/rtsp-server/rtsp-server.h:
9638 server: remove accept_client vmethod
9639 This vmethod is not very useful so remove it.
9641 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 * gst/rtsp-server/rtsp-server.c:
9644 server: don't crash on NULL GError
9646 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
9648 * gst/rtsp-server/rtsp-session-pool.c:
9649 rtsp-session-pool: corrected session timeout detection
9650 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
9652 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9654 * gst/rtsp-server/rtsp-client.c:
9655 client: improve debug
9657 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9659 * gst/rtsp-server/rtsp-client.c:
9660 * gst/rtsp-server/rtsp-client.h:
9661 * gst/rtsp-server/rtsp-server.c:
9662 server: refactor connection setup
9663 Let the server accept the socket connection and construct a GstRTSPConnection
9664 from it. Remove the code from the client and let the client only deal with
9665 a fully configure GstRTSPConnection object.
9666 We will need this later when the server will configure the connection for
9669 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * gst/rtsp-server/rtsp-stream.c:
9672 stream: keep the transport object alive
9673 Keep the transport object alive while we have it as qdata on the
9676 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
9678 * gst/rtsp-server/rtsp-client.c:
9679 * gst/rtsp-server/rtsp-server.c:
9680 rtsp-server: Do not crash on nmapping of server
9681 * generate error when gst_rtsp_connection_accept fails
9682 * do not stop accepting incoming connections because
9683 accepting a client fails
9684 https://bugzilla.gnome.org/show_bug.cgi?id=701072
9686 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
9688 * gst/rtsp-server/rtsp-client.c:
9689 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
9690 https://bugzilla.gnome.org/show_bug.cgi?id=700953
9692 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9694 * gst/rtsp-server/rtsp-sdp.c:
9695 rtsp-sdp: Parse framerate caps field and set SDP attribute
9696 The SDP attribute and its format is described in RFC4566.
9697 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9699 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
9701 * gst/rtsp-server/rtsp-sdp.c:
9702 rtsp-sdp: Parse width/height from caps and set SDP attribute
9703 The SDP attribute and its format is described in RFC6064.
9704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9706 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
9708 * gst/rtsp-server/rtsp-sdp.c:
9709 * tests/check/gst/client.c:
9710 rtsp-sdp: add bandwidth line
9711 https://bugzilla.gnome.org/show_bug.cgi?id=699220
9713 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9716 Automatic update of common submodule
9717 From 5edcd85 to 098c0d7
9719 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9721 * tests/check/gst/media.c:
9722 tests: add dynamic payloader prepare/unprepare check
9724 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9726 * gst/rtsp-server/rtsp-media.c:
9727 media: release lock when removing fakesink
9729 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9731 * gst/rtsp-server/rtsp-stream.c:
9732 stream: set elements to NULL before removing
9733 When removing a stream, set the elements to NULL first. This avoids
9734 element-is-not-in-NULL-state errors when we dispose the elements.
9736 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9739 Automatic update of common submodule
9740 From 3cb3d3c to 5edcd85
9742 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9744 * gst/rtsp-server/rtsp-media.c:
9745 * gst/rtsp-server/rtsp-media.h:
9746 media: listen to pad-removed signals
9747 Listen to the pad-removed signal and remove the stream associated with the
9749 Add signal to be notified of the removed pad.
9750 Remove the fakesink in unprepare()
9751 Fix signatures of the signal methods
9753 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9755 * examples/test-sdp.c:
9756 tests: add example of reusable pipelines
9758 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
9760 * gst/rtsp-server/rtsp-stream.c:
9761 * gst/rtsp-server/rtsp-stream.h:
9762 stream: add method to get the srcpad
9764 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9766 * tests/check/gst/media.c:
9767 check: add media prepare/unprepare test
9768 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9770 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
9772 * gst/rtsp-server/rtsp-media.c:
9773 media: disconnect from signal handlers in unprepare()
9774 We connected to the pad-added and no-more-pads signals in prepare() so
9775 we need to disconnect from them in unprepare().
9776 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9778 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
9780 * gst/rtsp-server/rtsp-media.c:
9781 media: don't free streams array
9782 Don't free the streams array in the unprepare() method, they were not
9784 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9786 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
9788 * gst/rtsp-server/rtsp-media.c:
9789 media: don't unref the pipeline in unprepare
9790 Unprepare() should undo what prepare() does. Because the pipeline is
9791 not created in prepare(), we should not unref it in unprepare()
9793 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
9795 * gst/rtsp-server/rtsp-stream.c:
9796 stream: clear session and caps for reuse
9797 Set the session and caps to NULL after unref otherwise we might unref
9799 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9801 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
9803 * gst/rtsp-server/rtsp-client.c:
9804 client: send out teardown signal before tearing down
9805 The advantage is that in the signal handler you get direct access to
9806 information about what streams are about to get torn down (in the
9807 GstRTSPClientState).
9808 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
9810 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
9812 * gst/rtsp-server/rtsp-client.c:
9813 * gst/rtsp-server/rtsp-client.h:
9814 client: expose connection
9815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
9817 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
9820 Automatic update of common submodule
9821 From aed87ae to 3cb3d3c
9823 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9825 * gst/rtsp-server/rtsp-media.c:
9826 * gst/rtsp-server/rtsp-media.h:
9827 * gst/rtsp-server/rtsp-session-media.c:
9828 * gst/rtsp-server/rtsp-session-media.h:
9829 media: add method to get the base_time of the pipeline
9830 Together with a shared clock, this base-time could eventually be sent to
9831 the client so that it can reconstruct the exact running-time of the clock
9834 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9836 * gst/rtsp-server/Makefile.am:
9837 * gst/rtsp-server/rtsp-media.c:
9838 * gst/rtsp-server/rtsp-media.h:
9839 * gst/rtsp-server/rtsp-sdp.c:
9840 media: add GstNetTimeProvider support
9841 Add a property to let the media provide a GstNetTimeProvider for its clock.
9842 Make methods to get the clock and nettimeprovider
9843 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
9844 provider and also the current time of the clock. This should make it possible
9845 for (GStreamer) clients to slave their clock to the server clock.
9847 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
9850 Automatic update of common submodule
9851 From 04c7a1e to aed87ae
9853 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9855 * gst/rtsp-server/rtsp-media.c:
9856 media: wait for buffering to complete
9857 Wait for buffering to complete before changing the state to the target state.
9859 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9861 * gst/rtsp-server/rtsp-media.c:
9862 media: small cleanup
9864 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
9866 * tests/check/gst/rtspserver.c:
9867 tests: remove extra unref in test_setup_non_existing_stream
9868 The unref is not needed anymore, teardown runs without it.
9869 https://bugzilla.gnome.org/show_bug.cgi?id=696542
9871 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
9873 * tests/check/gst/rtspserver.c:
9874 tests: GSocketService cleanup in test_bind_already_in_use
9875 Use g_socket_service_stop so the rtspserver test stops listening for
9876 incoming connections in test_bind_already_in_use.
9877 https://bugzilla.gnome.org/show_bug.cgi?id=696541
9879 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
9881 * gst/rtsp-server/rtsp-media-factory.c:
9882 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
9883 Instead use a GWeakRef which is safe to use
9884 This is a known GLib bug, see:
9885 https://bugzilla.gnome.org/show_bug.cgi?id=667145
9887 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
9889 * gst/rtsp-server/rtsp-client.c:
9890 * gst/rtsp-server/rtsp-media.c:
9891 * gst/rtsp-server/rtsp-media.h:
9892 * gst/rtsp-server/rtsp-sdp.c:
9893 * tests/check/gst/media.c:
9894 * tests/check/gst/rtspserver.c:
9895 rtsp-media/client: Reply to PLAY request with same type of Range
9896 Remember the type of Range from the PLAY request and use the same type for
9899 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
9901 * gst/rtsp-server/rtsp-client.c:
9902 * gst/rtsp-server/rtsp-client.h:
9903 * tests/check/gst/client.c:
9904 rtsp-client: expose uri
9906 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
9908 * tests/check/gst/mediafactory.c:
9909 tests: Hold ref while creating second media
9910 To test if the media aren't shared, make sure we keep the first one while creating a second
9911 otherwise the same memory address may be reused.
9913 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
9916 configure: remove out-of-date comment
9918 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
9921 .gitignore: ignore more build files
9923 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
9925 * tests/check/Makefile.am:
9926 tests: use right _LIBS variable for gst-plugins-base libs
9928 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9930 * tests/check/Makefile.am:
9931 check: add librtp to libs
9933 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
9935 * tests/check/gst/rtspserver.c:
9936 tests: Add test to check selecting a port the server will send from
9938 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
9940 * tests/check/gst/rtspserver.c:
9941 tests: Make sure packets are actually received
9943 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9945 * gst/rtsp-server/rtsp-stream.c:
9946 stream: Select unicast address from pool if appropriate
9948 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
9950 * gst/rtsp-server/rtsp-stream.c:
9951 stream: Properties are always there in Gst 1.0
9953 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9955 * tests/check/gst/addresspool.c:
9956 tests: Add tests for unicast addresses in pool
9958 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
9960 * gst/rtsp-server/rtsp-address-pool.c:
9961 * tests/check/gst/addresspool.c:
9962 address-pool: Verify that multicast addresses are used for multicast and vice-versa
9964 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
9966 * docs/libs/gst-rtsp-server-sections.txt:
9967 * gst/rtsp-server/rtsp-address-pool.c:
9968 * gst/rtsp-server/rtsp-address-pool.h:
9969 * gst/rtsp-server/rtsp-stream.c:
9970 * tests/check/gst/addresspool.c:
9971 address-pool: Add unicast addresses
9973 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9976 * gst/rtsp-server/rtsp-server.c:
9977 * tests/check/gst/rtspserver.c:
9978 rtsp-server: Limit the number of threads per server instance
9979 If we exceed the maximum, just round robin the clients over the existing
9982 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
9984 * gst/rtsp-server/rtsp-server.c:
9985 rtsp-server: No need to store the GMainContext in the client context
9987 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
9989 * tests/check/gst/rtspserver.c:
9990 tests: Add test for client disconnection
9992 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9994 * tests/check/gst/rtspserver.c:
9995 tests: Test client and session timeouts with multiple threads
9997 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
9999 * gst/rtsp-server/rtsp-address-pool.c:
10000 * gst/rtsp-server/rtsp-auth.c:
10001 * gst/rtsp-server/rtsp-client.c:
10002 * gst/rtsp-server/rtsp-media-factory-uri.c:
10003 * gst/rtsp-server/rtsp-media-factory.c:
10004 * gst/rtsp-server/rtsp-media.c:
10005 * gst/rtsp-server/rtsp-mount-points.c:
10006 * gst/rtsp-server/rtsp-server.c:
10007 * gst/rtsp-server/rtsp-session-media.c:
10008 * gst/rtsp-server/rtsp-session-pool.c:
10009 * gst/rtsp-server/rtsp-session.c:
10010 Document locking and its order
10012 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10014 * tests/check/gst/rtspserver.c:
10015 tests: Test that slow DESCRIBE don't block other clients
10017 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10019 * tests/check/gst/client.c:
10020 tests: Add tests for client-requested multicast address
10022 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10024 * docs/libs/gst-rtsp-server-sections.txt:
10025 docs: Put the various functions in the right sections
10027 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10029 * docs/libs/gst-rtsp-server-docs.sgml:
10030 * docs/libs/gst-rtsp-server-sections.txt:
10031 * gst/rtsp-server/rtsp-address-pool.c:
10032 * gst/rtsp-server/rtsp-address-pool.h:
10033 docs: Generate docs for GstRTSPAddressPool
10035 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10037 * gst/rtsp-server/rtsp-client.c:
10038 * gst/rtsp-server/rtsp-stream.c:
10039 * gst/rtsp-server/rtsp-stream.h:
10040 client: Check client provided addresses against the address pool
10042 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10044 * gst/rtsp-server/rtsp-address-pool.c:
10045 * gst/rtsp-server/rtsp-address-pool.h:
10046 * tests/check/gst/addresspool.c:
10047 address-pool: Add API to request a specific address from the pool
10048 Also add relevant unit tests.
10050 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10052 * tests/check/gst/mediafactory.c:
10053 tests: Check the passing around of a RTSPAddressPool
10054 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10055 way down to the stream.
10057 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10059 * tests/check/gst/addresspool.c:
10060 tests: Add more tests for the address pool
10062 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10064 * gst/rtsp-server/rtsp-address-pool.c:
10065 address-pool: Fix off by one error
10066 When splitting a port range, the port after a skip is not part of range.
10068 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10071 Automatic update of common submodule
10072 From 2de221c to 04c7a1e
10074 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10077 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10078 AM_CONFIG_HEADER was removed in automake 1.13
10079 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10081 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10084 Automatic update of common submodule
10085 From a942293 to 2de221c
10087 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10089 * gst/rtsp-server/rtsp-client.c:
10090 client: make sure the watch exists while sending data
10091 Protect the send_func with a lock. This allows us to wait for sending
10092 to complete before changing the send_func and user_data. We add an
10093 extra ref to the watch to make sure that it remains valid during
10095 When closing the connection, set the send_func to NULL
10096 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10098 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10100 * tests/check/Makefile.am:
10101 tests: use GST_*_1_0 environment variables everywhere
10102 The _1_0 suffixed environment variables override the
10103 non-suffixed ones, so if we're in an environment that
10104 sets the _1_0 suffixed ones, such as jhbuild, we need
10105 to set those to make sure ours actually always get
10108 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10111 Automatic update of common submodule
10112 From acb04d9 to a942293
10114 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10116 * gst/rtsp-server/rtsp-client.c:
10117 rtsp-client: set the client backlog
10118 Set the client backlog to a reasonable default
10120 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10122 * gst/rtsp-server/rtsp-media.c:
10123 rtsp-media: Make the element a constructor parameter
10124 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10126 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10128 * docs/libs/Makefile.am:
10129 docs: Link with gcov library when gcov is enabled
10130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10132 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10134 * gst/rtsp-server/rtsp-media.c:
10135 media: match prepare with unprepare
10136 Really unprepare when there were an equal amount of prepare calls.
10138 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10140 * gst/rtsp-server/rtsp-media.c:
10141 media: media has to be unprepared in finalize
10142 Because unprepare takes away the last ref on the media.
10144 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10146 * gst/rtsp-server/rtsp-client.c:
10147 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10148 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10149 We can't use the refcount to trigger unprepare because it is the unprepare call
10150 that removes the last refcount after all messages are consumed. What we should
10151 probably do is make a prepared refcount and only unprepare when the refcount
10154 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10156 * gst/rtsp-server/rtsp-media.c:
10157 media: let the source unref the last media ref
10158 the last ref to the media is held by the source so we don't need to add more ref
10159 and unrefs, we simply destroy the media when the source is gone.
10161 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10163 * gst/rtsp-server/rtsp-media.c:
10164 media: improve debug
10166 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10168 * gst/rtsp-server/rtsp-media.c:
10170 Make sure we are in the right state when collecting the position and duration.
10171 Only make ourselves PREPARED when we were previously PREPARING.
10173 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10175 * gst/rtsp-server/rtsp-media.c:
10176 media: use g_object_ref/unref for GObjects
10178 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10180 * gst/rtsp-server/rtsp-client.c:
10181 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10182 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10183 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10184 isn't being used anymore.
10186 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10188 * gst/rtsp-server/rtsp-media.c:
10189 Fix compiler warning
10191 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10193 * gst/rtsp-server/rtsp-media-factory-uri.c:
10194 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10196 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10198 * gst/rtsp-server/rtsp-session-media.h:
10201 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10203 * gst/rtsp-server/rtsp-media.c:
10204 * tests/check/gst/media.c:
10205 media: avoid element leak
10207 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10209 * gst/rtsp-server/rtsp-media.c:
10210 media: require an element in media constructor
10212 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10214 * gst/rtsp-server/rtsp-client.c:
10215 Revert "client: TEARDOWN brings that state to Init again"
10216 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10217 The object is already disposed, there is no point in setting the state.
10219 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10221 * gst/rtsp-server/rtsp-client.c:
10222 client: TEARDOWN brings that state to Init again
10224 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10226 * docs/libs/gst-rtsp-server-sections.txt:
10227 * examples/test-auth.c:
10228 * gst/rtsp-server/rtsp-auth.c:
10229 * gst/rtsp-server/rtsp-auth.h:
10230 * gst/rtsp-server/rtsp-client.c:
10231 * gst/rtsp-server/rtsp-client.h:
10232 * gst/rtsp-server/rtsp-media-factory-uri.c:
10233 * gst/rtsp-server/rtsp-media-factory-uri.h:
10234 * gst/rtsp-server/rtsp-media-factory.c:
10235 * gst/rtsp-server/rtsp-media-factory.h:
10236 * gst/rtsp-server/rtsp-media.c:
10237 * gst/rtsp-server/rtsp-media.h:
10238 * gst/rtsp-server/rtsp-mount-points.c:
10239 * gst/rtsp-server/rtsp-mount-points.h:
10240 * gst/rtsp-server/rtsp-sdp.c:
10241 * gst/rtsp-server/rtsp-server.c:
10242 * gst/rtsp-server/rtsp-server.h:
10243 * gst/rtsp-server/rtsp-session-media.c:
10244 * gst/rtsp-server/rtsp-session-media.h:
10245 * gst/rtsp-server/rtsp-session-pool.c:
10246 * gst/rtsp-server/rtsp-session-pool.h:
10247 * gst/rtsp-server/rtsp-session.c:
10248 * gst/rtsp-server/rtsp-session.h:
10249 * gst/rtsp-server/rtsp-stream-transport.c:
10250 * gst/rtsp-server/rtsp-stream-transport.h:
10251 * gst/rtsp-server/rtsp-stream.c:
10252 * gst/rtsp-server/rtsp-stream.h:
10253 * tests/check/gst/media.c:
10254 rtsp: make object details private
10255 Make all object details private
10256 Add methods to access private bits
10258 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10260 * tests/check/Makefile.am:
10261 * tests/check/gst/media.c:
10262 tests: add media tests
10264 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10266 * gst/rtsp-server/rtsp-media.c:
10267 media: check if prepared for some methods
10268 Check that the media object is prepared before doing seek and getting the
10269 current position etc.
10270 Add some g_return checks.
10272 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10274 * tests/check/Makefile.am:
10275 * tests/check/gst/mediafactory.c:
10276 tests: add mediafactory test
10278 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10280 * gst/rtsp-server/rtsp-stream.c:
10281 stream: improve debug
10283 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10285 * gst/rtsp-server/rtsp-media.c:
10286 * gst/rtsp-server/rtsp-media.h:
10287 media: unref pipeline in finalize to avoid leaking it
10289 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10291 * gst/rtsp-server/rtsp-media-factory-uri.c:
10292 * gst/rtsp-server/rtsp-media.c:
10293 rtsp: use gst_object_unref on GstObjects
10295 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10297 * gst/rtsp-server/rtsp-media-factory.c:
10298 media-factory: require an url
10300 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10302 * examples/test-uri.c:
10303 examples: fix include
10305 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10307 * gst/rtsp-server/rtsp-server.h:
10308 server: remove unused include
10310 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10312 * tests/check/Makefile.am:
10313 * tests/check/gst/mountpoints.c:
10314 tests: add test for mountpoints
10316 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10318 * gst/rtsp-server/rtsp-client.c:
10319 client: fix factory leak
10320 Keep the factory in the state object only for authorization checks and make
10321 sure we unref it on failure. Also don't keep invalid objects in the state
10324 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10326 * gst/rtsp-server/rtsp-mount-points.c:
10327 mounts: add g_return_if guards
10329 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10331 * tests/check/gst/client.c:
10332 tests: add more tests
10334 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10336 * gst/rtsp-server/rtsp-client.c:
10337 client: improve debug
10339 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10341 * gst/rtsp-server/rtsp-client.c:
10342 client: improve debug and fix leaks
10343 Cleanup the uri and session when there is a bad request.
10345 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10350 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10352 * tests/check/gst/client.c:
10353 test: add test for session in options request
10355 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10357 * gst/rtsp-server/rtsp-client.c:
10358 client: use 454 when session can't be found
10359 We should use 454 when a session can't be found because there was no session
10360 pool configured in the server. This is not a server configuration problem
10361 because the server on which the request is done might not be the same one that
10362 will keep the sessions for us and so it does not need to support sessions.
10364 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10366 * gst/rtsp-server/rtsp-client.c:
10367 client: only free connection when there is one
10368 It's possible that the client doesn't have a connection when we try to free it.
10370 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10372 * tests/check/Makefile.am:
10373 * tests/check/gst/client.c:
10374 tests: add unit test for the client object
10376 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10378 * gst/rtsp-server/rtsp-client.c:
10379 client: small cleanup
10381 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10383 * gst/rtsp-server/rtsp-client.h:
10384 client: remove unused include
10386 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10388 * gst/rtsp-server/rtsp-client.c:
10389 client: fix compilation
10391 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10393 * gst/rtsp-server/rtsp-client.c:
10394 client: call destroy without the lock
10396 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10398 * gst/rtsp-server/rtsp-client.c:
10399 * gst/rtsp-server/rtsp-client.h:
10400 client: make the client usable without a socket
10401 Make a method to let the client handle a message and a callback when the client
10402 wants us to send a response message back. This makes it possible to also use the
10403 client object without the sockets, which should make it easier to test.
10405 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10407 * gst/rtsp-server/rtsp-client.c:
10408 * gst/rtsp-server/rtsp-client.h:
10409 client: small cleanup
10411 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10413 * docs/libs/gst-rtsp-server-sections.txt:
10414 * gst/rtsp-server/rtsp-client.c:
10415 * gst/rtsp-server/rtsp-client.h:
10416 * gst/rtsp-server/rtsp-server.c:
10417 client: remove reference to server
10418 We don't need to keep a ref to the server
10420 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10422 * gst/rtsp-server/rtsp-client.c:
10423 * gst/rtsp-server/rtsp-client.h:
10424 client: add locking
10425 Also add some g_return_if()
10427 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10429 * gst/rtsp-server/rtsp-client.c:
10430 client: log more errors
10432 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10434 * gst/rtsp-server/rtsp-client.c:
10435 client: fix compilation
10437 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10439 * gst/rtsp-server/rtsp-client.c:
10440 * gst/rtsp-server/rtsp-client.h:
10441 client: add generic close-after-send support
10442 Add a property to send_response() to close the connection after the response has
10443 been sent to the client.
10445 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10448 * docs/libs/gst-rtsp-server-docs.sgml:
10449 * docs/libs/gst-rtsp-server-sections.txt:
10450 * docs/libs/gst-rtsp-server.types:
10451 * examples/test-auth.c:
10452 * examples/test-launch.c:
10453 * examples/test-mp4.c:
10454 * examples/test-multicast.c:
10455 * examples/test-multicast2.c:
10456 * examples/test-ogg.c:
10457 * examples/test-readme.c:
10458 * examples/test-sdp.c:
10459 * examples/test-uri.c:
10460 * examples/test-video.c:
10461 * gst/rtsp-server/Makefile.am:
10462 * gst/rtsp-server/rtsp-auth.h:
10463 * gst/rtsp-server/rtsp-client.c:
10464 * gst/rtsp-server/rtsp-client.h:
10465 * gst/rtsp-server/rtsp-media-mapping.c:
10466 * gst/rtsp-server/rtsp-media-mapping.h:
10467 * gst/rtsp-server/rtsp-mount-points.c:
10468 * gst/rtsp-server/rtsp-mount-points.h:
10469 * gst/rtsp-server/rtsp-server.c:
10470 * gst/rtsp-server/rtsp-server.h:
10471 * gst/rtsp-server/rtsp-session-media.c:
10472 * gst/rtsp-server/rtsp-session-pool.c:
10473 * gst/rtsp-server/rtsp-session-pool.h:
10474 * tests/check/gst/rtspserver.c:
10475 MediaMapping -> MountPoints
10476 Describes better what the object manages.
10478 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10481 configure: bump required version of -base
10483 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10485 * gst/rtsp-server/rtsp-media.c:
10488 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10490 * gst/rtsp-server/rtsp-media.c:
10491 * gst/rtsp-server/rtsp-media.h:
10492 media: support more Range formats
10493 Use the new -base methods to convert the Range string into a seek start and stop
10496 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10498 * examples/test-launch.c:
10499 examples: fix whitespace
10501 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10503 * examples/test-auth.c:
10504 test-auth: add example of how to remove sessions
10505 Add an example of the session filter api.
10507 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10509 * examples/test-uri.c:
10510 test-uri: remove mapping example
10512 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10514 * examples/test-uri.c:
10515 test-uri: fix callback signature
10517 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10519 * gst/rtsp-server/rtsp-media-factory.c:
10520 factory: keep ref to factory while media active
10521 While the media from a factory is alive, keep a ref to the factory.
10522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10524 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10526 * gst/rtsp-server/rtsp-media-factory-uri.c:
10527 factory-uri: add some debug
10529 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10531 * gst/rtsp-server/rtsp-stream.c:
10532 stream: set udp sources to PLAYING
10533 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10534 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10536 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10538 * gst/rtsp-server/rtsp-media-factory-uri.c:
10539 factory-uri: take ref to factory
10540 Take a ref to the factory that we place in our list.
10542 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10544 * tests/Makefile.am:
10545 * tests/test-reuse.c:
10546 test: add test for server reuse
10547 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10549 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10551 * gst/rtsp-server/rtsp-server.c:
10552 server: start and stop multiple times
10553 Stop listening on the RTSP port when the GSource is removed, so clients
10554 can't connect and the server can be started again.
10555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10557 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10559 * gst/rtsp-server/rtsp-server.c:
10560 server: fix small leak
10562 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10564 * gst/rtsp-server/rtsp-media.c:
10565 media: unref source in finish_unprepare
10566 The source is created in prepare, unref it in finish_unprepare.
10567 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10569 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10571 * gst/rtsp-server/rtsp-client.c:
10572 * gst/rtsp-server/rtsp-media.c:
10573 rtsp-media: remove bus watch before finalizing
10574 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10575 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10576 the GDestroyNotify function.
10577 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10578 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10579 gst_rtsp_media_unprepare before unreffing the media.
10580 This way, the bus watch will be removed before the media is finalized.
10581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10583 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10585 * gst/rtsp-server/rtsp-client.c:
10586 * gst/rtsp-server/rtsp-client.h:
10587 client: wait until the TEARDOWN response is sent to close the connection
10588 Responses can be sent async so we need to wait until the TEARDOWN response has
10589 been written before we close the connection to the client. This avoids the risk
10590 of writing/polling closed sockets.
10591 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10593 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10595 * gst/rtsp-server/rtsp-stream.c:
10596 rtsp-stream: plug socket leak
10597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10599 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10602 Automatic update of common submodule
10603 From 6bb6951 to a72faea
10605 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10607 * gst/rtsp-server/rtsp-media-factory-uri.c:
10608 rtsp-server: don't use deprecated API
10610 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10612 * gst/rtsp-server/rtsp-client.c:
10613 rtsp-client: fix unused-but-set-variable compiler warning
10614 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10616 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10619 * docs/libs/gst-rtsp-server-sections.txt:
10620 * gst/rtsp-server/rtsp-client.c:
10623 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10625 * examples/Makefile.am:
10626 * examples/test-multicast2.c:
10627 examples: add another multicast example
10628 Add an example for how to configure separate multicast ranges for each media
10631 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10633 * examples/test-multicast.c:
10636 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10638 * gst/rtsp-server/rtsp-client.c:
10639 * gst/rtsp-server/rtsp-media.c:
10640 * gst/rtsp-server/rtsp-session-media.c:
10641 * gst/rtsp-server/rtsp-session-media.h:
10642 * gst/rtsp-server/rtsp-stream-transport.c:
10643 * gst/rtsp-server/rtsp-stream-transport.h:
10644 stream: use the address managed by the stream
10645 Use the address managed by the stream for multicast. This allows us to have 1
10646 multicast address for each stream.
10647 Because the address is now managed by the stream we don't have to pass it around
10649 Set the address pool on the streams.
10651 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10653 * gst/rtsp-server/rtsp-client.c:
10654 * gst/rtsp-server/rtsp-media.c:
10655 * gst/rtsp-server/rtsp-stream.c:
10656 rtsp: improve debug
10658 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10660 * gst/rtsp-server/rtsp-media.c:
10661 * gst/rtsp-server/rtsp-media.h:
10662 media: add signal for new streams
10663 This allows applications to listen for new streams and configure properties on
10664 them, like the address pool.
10666 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10668 * gst/rtsp-server/rtsp-media.c:
10669 media: configure address pool in new streams
10671 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10673 * gst/rtsp-server/rtsp-stream.c:
10674 * gst/rtsp-server/rtsp-stream.h:
10675 stream: add methods to deal with address pool
10676 Add methods to get and set the address pool for the stream
10677 Add method to allocate and get the multicast addresses for this stream.
10679 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10681 * docs/libs/gst-rtsp-server-sections.txt:
10682 * gst/rtsp-server/rtsp-media.c:
10683 * gst/rtsp-server/rtsp-media.h:
10684 media: remove MTU property
10685 It is a stream property
10687 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10689 * gst/rtsp-server/rtsp-client.c:
10690 client: set blocksize only on stream
10691 Set the blocksize only on the current stream.
10693 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10695 * gst/rtsp-server/rtsp-stream.c:
10696 stream: share src and sink sockets
10697 the allocated socket is in the used-socket property, not socket.
10699 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10701 * gst/rtsp-server/rtsp-address-pool.c:
10702 * gst/rtsp-server/rtsp-address-pool.h:
10703 * gst/rtsp-server/rtsp-client.c:
10704 * gst/rtsp-server/rtsp-session-media.c:
10705 * gst/rtsp-server/rtsp-session-media.h:
10706 * gst/rtsp-server/rtsp-stream-transport.c:
10707 * gst/rtsp-server/rtsp-stream-transport.h:
10708 * tests/check/gst/addresspool.c:
10709 rtsp: make address-pool return an address object
10710 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
10711 store more info in the structure and allows us to more easily return the address
10712 to the right pool when no longer needed.
10713 Pass the address to the StreamTransport so that we can return it to the pool
10714 when the stream transport is freed or changed.
10716 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10718 * examples/Makefile.am:
10719 * examples/test-multicast.c:
10720 examples: add multicast example
10721 Show how to set up the multicast address pool so that media can be
10722 server with multicast.
10724 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10726 * gst/rtsp-server/rtsp-client.c:
10727 * gst/rtsp-server/rtsp-media-factory.c:
10728 * gst/rtsp-server/rtsp-media-factory.h:
10729 * gst/rtsp-server/rtsp-media.c:
10730 * gst/rtsp-server/rtsp-media.h:
10731 rtsp: use AddressPool
10732 Remove the multicast_group property.
10733 Use the configured addresspool to allocate multicast addresses.
10735 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10737 * gst/rtsp-server/rtsp-address-pool.c:
10738 * gst/rtsp-server/rtsp-address-pool.h:
10739 address-pool: add clear method
10741 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10743 * gst/rtsp-server/rtsp-address-pool.c:
10744 address-pool: small cleanups
10746 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10748 * tests/check/Makefile.am:
10749 * tests/check/gst/addresspool.c:
10750 tests: add addresspool unit test
10752 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10754 * gst/rtsp-server/Makefile.am:
10755 * gst/rtsp-server/rtsp-address-pool.c:
10756 * gst/rtsp-server/rtsp-address-pool.h:
10757 address-pool: add object to manage multicast addresses
10758 Make an object that can manage a rage of multicast addresses and ports.
10760 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10762 * gst/rtsp-server/rtsp-server.c:
10763 server: set default max-threads property
10765 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10767 * gst/rtsp-server/rtsp-media.c:
10768 media: wait for concurrent _prepare
10769 If a prepare is busy, wait for the result.
10771 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10773 * gst/rtsp-server/rtsp-media.c:
10774 media: add lock around message handler
10775 We don't want to dispatch messages while we are still processing the result of
10778 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10780 * gst/rtsp-server/rtsp-media.c:
10781 * gst/rtsp-server/rtsp-media.h:
10782 media: add lock to protect state changes
10784 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10786 * gst/rtsp-server/rtsp-stream.c:
10787 * gst/rtsp-server/rtsp-stream.h:
10788 stream: add locking
10790 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10792 * gst/rtsp-server/rtsp-stream-transport.c:
10793 * gst/rtsp-server/rtsp-stream-transport.h:
10794 * gst/rtsp-server/rtsp-stream.c:
10795 stream-transport: add keep-alive method
10797 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10799 * gst/rtsp-server/rtsp-stream-transport.c:
10800 * gst/rtsp-server/rtsp-stream-transport.h:
10801 * gst/rtsp-server/rtsp-stream.c:
10802 stream-transport: add method to handle RTP/RTCP
10803 Call new methods instead of poking into the structures directly.
10805 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10807 * gst/rtsp-server/rtsp-session-media.c:
10808 * gst/rtsp-server/rtsp-session-media.h:
10809 session-media: add locking
10811 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10813 * gst/rtsp-server/rtsp-session.c:
10814 * gst/rtsp-server/rtsp-session.h:
10815 session: add locking
10817 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10819 * gst/rtsp-server/rtsp-server.c:
10820 server: free old socket
10822 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10824 * gst/rtsp-server/rtsp-media-mapping.c:
10825 * gst/rtsp-server/rtsp-media-mapping.h:
10826 mapping: add locking
10828 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10830 * gst/rtsp-server/rtsp-media-factory.c:
10831 media-factory: add locking
10833 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10835 * gst/rtsp-server/rtsp-auth.c:
10836 * gst/rtsp-server/rtsp-auth.h:
10839 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10841 * gst/rtsp-server/rtsp-server.c:
10842 * gst/rtsp-server/rtsp-server.h:
10843 server: add max-thread property
10845 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10847 * gst/rtsp-server/rtsp-server.c:
10848 * gst/rtsp-server/rtsp-server.h:
10849 server: use a threadpool for the mainloops
10851 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10853 * gst/rtsp-server/rtsp-client.c:
10854 * gst/rtsp-server/rtsp-client.h:
10855 client: rename method
10856 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
10857 don't really create the client from the socket, we use the socket for the
10860 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10862 * gst/rtsp-server/rtsp-client.c:
10863 * gst/rtsp-server/rtsp-client.h:
10864 * gst/rtsp-server/rtsp-server.c:
10865 server: rework maincontext handling in clients
10866 Make a separate method to attach a client to a MainContext.
10867 Let the server decide in what GMainContext the client will operate and give this
10868 context to the client in attach. Then the server can later decide to use a
10869 separate thread for each client or just use the mainthread.
10871 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10873 * gst/rtsp-server/rtsp-client.c:
10874 * gst/rtsp-server/rtsp-session.c:
10875 * gst/rtsp-server/rtsp-session.h:
10876 session: move session header code in session object
10878 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10882 * examples/test-auth.c:
10883 * examples/test-launch.c:
10884 * examples/test-mp4.c:
10885 * examples/test-ogg.c:
10886 * examples/test-readme.c:
10887 * examples/test-sdp.c:
10888 * examples/test-uri.c:
10889 * examples/test-video.c:
10890 * gst/rtsp-server/rtsp-auth.c:
10891 * gst/rtsp-server/rtsp-auth.h:
10892 * gst/rtsp-server/rtsp-client.c:
10893 * gst/rtsp-server/rtsp-client.h:
10894 * gst/rtsp-server/rtsp-media-factory-uri.c:
10895 * gst/rtsp-server/rtsp-media-factory-uri.h:
10896 * gst/rtsp-server/rtsp-media-factory.c:
10897 * gst/rtsp-server/rtsp-media-factory.h:
10898 * gst/rtsp-server/rtsp-media-mapping.c:
10899 * gst/rtsp-server/rtsp-media-mapping.h:
10900 * gst/rtsp-server/rtsp-media.c:
10901 * gst/rtsp-server/rtsp-media.h:
10902 * gst/rtsp-server/rtsp-params.c:
10903 * gst/rtsp-server/rtsp-params.h:
10904 * gst/rtsp-server/rtsp-sdp.c:
10905 * gst/rtsp-server/rtsp-sdp.h:
10906 * gst/rtsp-server/rtsp-server.c:
10907 * gst/rtsp-server/rtsp-server.h:
10908 * gst/rtsp-server/rtsp-session-media.c:
10909 * gst/rtsp-server/rtsp-session-media.h:
10910 * gst/rtsp-server/rtsp-session-pool.c:
10911 * gst/rtsp-server/rtsp-session-pool.h:
10912 * gst/rtsp-server/rtsp-session.c:
10913 * gst/rtsp-server/rtsp-session.h:
10914 * gst/rtsp-server/rtsp-stream-transport.c:
10915 * gst/rtsp-server/rtsp-stream-transport.h:
10916 * gst/rtsp-server/rtsp-stream.c:
10917 * gst/rtsp-server/rtsp-stream.h:
10918 * tests/check/gst/rtspserver.c:
10919 * tests/test-cleanup.c:
10922 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10924 * gst/rtsp-server/rtsp-media.c:
10925 * gst/rtsp-server/rtsp-session-media.c:
10926 * gst/rtsp-server/rtsp-session.c:
10927 rtsp-server: added annotations to indicate type of ownership transfer of return values
10928 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10930 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
10933 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
10935 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10938 * bindings/Makefile.am:
10939 * bindings/vala/Makefile.am:
10940 * bindings/vala/gst-rtsp-server-0.10.deps:
10941 * bindings/vala/gst-rtsp-server-0.10.vapi:
10942 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10943 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10944 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10945 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10946 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10948 bindings: remove vala bindings
10949 They'll be reunited with the other GStreamer bindings
10950 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10952 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10954 * gst/rtsp-server/rtsp-client.c:
10955 * gst/rtsp-server/rtsp-session-media.c:
10956 * gst/rtsp-server/rtsp-session-media.h:
10957 * gst/rtsp-server/rtsp-stream-transport.c:
10958 * gst/rtsp-server/rtsp-stream-transport.h:
10959 rtsp: only create transport when needed
10960 Only create the StreamTransport when configured.
10962 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10964 * gst/rtsp-server/rtsp-client.c:
10965 client: small cleanup
10967 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10969 * gst/rtsp-server/rtsp-client.c:
10970 * gst/rtsp-server/rtsp-client.h:
10971 * gst/rtsp-server/rtsp-stream-transport.c:
10972 * gst/rtsp-server/rtsp-stream-transport.h:
10973 rtsp: refactor configuration of transport
10974 Move the configuration of the transport to a place where it makes
10977 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10979 * gst/rtsp-server/rtsp-client.c:
10980 client: refactor transport parsing
10982 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10984 * gst/rtsp-server/rtsp-client.c:
10985 client: refuse to change the MTU on shared media
10986 If we change the MTU of chared media, it changes for all clients.
10987 We don't want to set the MTU to something large for clients that
10990 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10992 * examples/test-mp4.c:
10993 * gst/rtsp-server/rtsp-media.c:
10994 small fixes to docs and debug
10996 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10998 * gst/rtsp-server/rtsp-stream.c:
10999 stream: transports must already have been removed
11001 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11003 * gst/rtsp-server/rtsp-media.c:
11004 * gst/rtsp-server/rtsp-stream.c:
11005 * gst/rtsp-server/rtsp-stream.h:
11006 stream: improve join and leave of the pipeline
11008 Do the cleanup properly
11011 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11013 * gst/rtsp-server/rtsp-media.c:
11014 media: move unprepare below default implementation
11015 Makes it easier to find the default implementation
11017 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11019 * gst/rtsp-server/rtsp-media.c:
11020 media: signal unprepared when we actually finish
11022 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11024 * gst/rtsp-server/rtsp-media.c:
11025 media: no need to unlock, unprepare does that when needed
11027 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11029 * docs/libs/gst-rtsp-server-sections.txt:
11030 * gst/rtsp-server/rtsp-media-factory.h:
11031 * gst/rtsp-server/rtsp-media-mapping.c:
11032 * gst/rtsp-server/rtsp-media.h:
11033 * gst/rtsp-server/rtsp-params.c:
11034 * gst/rtsp-server/rtsp-server.c:
11035 * gst/rtsp-server/rtsp-session-pool.h:
11036 * gst/rtsp-server/rtsp-session.c:
11037 * gst/rtsp-server/rtsp-session.h:
11038 * gst/rtsp-server/rtsp-stream-transport.h:
11039 * gst/rtsp-server/rtsp-stream.h:
11042 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11044 * gst/rtsp-server/rtsp-client.c:
11045 * gst/rtsp-server/rtsp-media-mapping.h:
11046 * gst/rtsp-server/rtsp-media.c:
11047 * gst/rtsp-server/rtsp-media.h:
11048 * gst/rtsp-server/rtsp-server.h:
11049 * gst/rtsp-server/rtsp-stream.c:
11050 * gst/rtsp-server/rtsp-stream.h:
11051 rtsp: fix MTU setting
11052 Fix setting of the MTU. There is no need for a vmethod.
11054 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11059 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11062 configure: bump version number after refactoring
11064 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11066 * gst/rtsp-server/Makefile.am:
11067 * gst/rtsp-server/rtsp-client.c:
11068 * gst/rtsp-server/rtsp-client.h:
11069 * gst/rtsp-server/rtsp-media-factory-uri.c:
11070 * gst/rtsp-server/rtsp-media-factory.c:
11071 * gst/rtsp-server/rtsp-media-factory.h:
11072 * gst/rtsp-server/rtsp-media.c:
11073 * gst/rtsp-server/rtsp-media.h:
11074 * gst/rtsp-server/rtsp-sdp.c:
11075 * gst/rtsp-server/rtsp-session-media.c:
11076 * gst/rtsp-server/rtsp-session-media.h:
11077 * gst/rtsp-server/rtsp-session.c:
11078 * gst/rtsp-server/rtsp-session.h:
11079 * gst/rtsp-server/rtsp-stream-transport.c:
11080 * gst/rtsp-server/rtsp-stream-transport.h:
11081 * gst/rtsp-server/rtsp-stream.c:
11082 * gst/rtsp-server/rtsp-stream.h:
11083 rtsp: massive refactoring
11084 Make GObjects from the remaining simple structures.
11085 Remove GstRTSPSessionStream, it's not needed.
11086 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11087 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11088 a GstRTSPStream should be transported to a client.
11089 Rename GstRTSPMediaFactory::get_element -> create_element because that
11090 more accurately describes what it does.
11091 Make nice methods instead of poking in the structures.
11092 Move some methods inside the relevant object source code.
11093 Use GPtrArray to store objects instead of plain arrays, it is more
11094 natural and allows us to more easily clean up.
11095 Move the allocation of udp ports to the Stream object. The Stream object
11096 contains the elements needed to stream the media to a client.
11097 Improve the prepare and unprepare methods. Unprepare should now undo
11098 everything prepare did. Improve also async unprepare when doing EOS on
11099 shutdown. Make sure we always unprepare correctly.
11101 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11103 * gst/rtsp-server/rtsp-client.c:
11104 rtsp-client: Unref server address clients connected to
11105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11107 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11109 * gst/rtsp-server/rtsp-server.c:
11110 rtsp-server: don't ref server socket if it is NULL
11111 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11112 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11114 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11116 * tests/check/Makefile.am:
11117 tests: Add libgio link dependency
11118 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11120 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11122 * gst/rtsp-server/rtsp-media-mapping.c:
11123 * gst/rtsp-server/rtsp-media-mapping.h:
11124 rtsp-media-mapping: rename find_media vfunc to find_factory
11125 The virtual method and class method should have the same name
11126 so it is correctly represented in GIR file
11127 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11129 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11131 * gst/rtsp-server/rtsp-auth.c:
11132 * gst/rtsp-server/rtsp-client.c:
11133 * gst/rtsp-server/rtsp-media-factory-uri.c:
11134 * gst/rtsp-server/rtsp-media-factory.c:
11135 * gst/rtsp-server/rtsp-media-mapping.c:
11136 * gst/rtsp-server/rtsp-media.c:
11137 * gst/rtsp-server/rtsp-server.c:
11138 * gst/rtsp-server/rtsp-session-pool.c:
11139 * gst/rtsp-server/rtsp-session.c:
11140 rtsp-server: fixed comments and GIR annotations
11141 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11143 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11145 * gst/rtsp-server/rtsp-media-mapping.c:
11146 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11148 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11150 * gst/rtsp-server/rtsp-server.c:
11151 rtsp-server: allow binding on port 0 (binds on a random port)
11153 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11155 * gst/rtsp-server/rtsp-server.c:
11156 * gst/rtsp-server/rtsp-server.h:
11157 rtsp-server: add bound-port property
11158 bound-port can be used to retrieve the port number when the server is bound on
11159 port 0, which binds on a random port.
11161 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11163 * gst/rtsp-server/rtsp-media-factory.c:
11164 * gst/rtsp-server/rtsp-media-factory.h:
11165 rtsp-media-factory: make ::get_element overridable by GI bindings
11166 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11167 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11168 as the invoker for ::get_element(), making it overridable by GI generated
11171 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11173 * gst/rtsp-server/rtsp-media-factory-uri.c:
11174 rtsp-media-factory-uri: don't autoplug parsers in a loop
11175 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11178 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11180 * gst/rtsp-server/Makefile.am:
11181 Explicitly link against gio. Fix link error on mac.
11183 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11185 * gst/rtsp-server/rtsp-session.c:
11186 session: add ttl to the transport header in SETUP
11187 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11189 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11191 * gst/rtsp-server/rtsp-client.c:
11192 * gst/rtsp-server/rtsp-client.h:
11193 * gst/rtsp-server/rtsp-media.c:
11194 client: Use client transport settings for multicast if allowed.
11195 This patch makes it possible for the client to send transport settings for
11196 multicast (destination && ttl). Client settings must be explicitly allowed or
11197 the server will use its own settings.
11198 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11200 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11203 Automatic update of common submodule
11204 From 6c0b52c to 6bb6951
11206 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11208 * gst/rtsp-server/rtsp-client.c:
11209 rtsp-client: do not destroy the rtsp watch
11210 Don't destroy the client watch while dispatching. The rtsp watch is
11211 automatically destroyed after the rtsp watch function closed() has
11213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11215 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11218 Automatic update of common submodule
11219 From 4f962f7 to 6c0b52c
11221 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11223 * gst/rtsp-server/rtsp-media.c:
11224 media: fix check for seekability
11226 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11228 * gst/rtsp-server/rtsp-client.c:
11229 client: use more GIO
11230 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11232 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11234 * gst/rtsp-server/rtsp-server.c:
11235 server: remove obsolete includes
11237 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11239 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11240 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11241 be available in "on_new_ssrc". The transports are added in
11242 gst_rtsp_media_set_state when going to PLAYING state. However,
11243 "on_new_ssrc" might be called before this happens.
11244 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11246 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11248 * gst/rtsp-server/rtsp-client.c:
11249 * gst/rtsp-server/rtsp-client.h:
11250 rtsp-client: add signals for rtsp requests (fixes #683287)
11252 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11254 * gst/rtsp-server/rtsp-client.c:
11255 * gst/rtsp-server/rtsp-client.h:
11256 add new-session signal to rtsp-client (fixes #683058)
11258 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11261 Automatic update of common submodule
11262 From 668acee to 4f962f7
11264 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11266 * gst/rtsp-server/rtsp-server.c:
11267 * tests/check/gst/rtspserver.c:
11268 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11269 Do not assume that *error is set in g_socket_address_enumerator_next.
11270 Added test_bind_already_in_use unit-test.
11271 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11273 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11276 Automatic update of common submodule
11277 From 94ccf4c to 668acee
11279 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11281 * gst/rtsp-server/rtsp-client.c:
11282 * gst/rtsp-server/rtsp-client.h:
11283 rtsp-client: make create_sdp virtual method
11284 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11286 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11289 Automatic update of common submodule
11290 From 98e386f to 94ccf4c
11292 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11294 * gst/rtsp-server/rtsp-client.c:
11297 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11299 * gst/rtsp-server/rtsp-client.c:
11300 * gst/rtsp-server/rtsp-client.h:
11301 * gst/rtsp-server/rtsp-server.c:
11302 * gst/rtsp-server/rtsp-server.h:
11303 rtsp-server: use an existing socket to establish HTTP tunnel
11304 Make it possible to transfer a socket from an HTTP server to be used as
11305 an RTSP over HTTP tunnel.
11307 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11309 * gst/rtsp-server/rtsp-client.c:
11310 * gst/rtsp-server/rtsp-media.c:
11311 * gst/rtsp-server/rtsp-media.h:
11312 rtsp: Handle the blocksize parameter
11313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11315 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11317 * tests/check/Makefile.am:
11318 * tests/check/gst/rtspserver.c:
11319 Have unit test get header from source dir, not installed dir
11320 This makes compilation of unit tests work in a build directory other
11321 than the source directory.
11322 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11324 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11326 * gst/rtsp-server/rtsp-media.c:
11327 rtsp-media: update for gst_element_make_from_uri() changes
11329 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11332 * tests/Makefile.am:
11333 * tests/check/Makefile.am:
11334 * tests/check/gst/rtspserver.c:
11335 rtsp: add unit test
11336 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11338 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11340 * gst/rtsp-server/rtsp-media.c:
11341 rtsp-media: don't collect media stats when going to NULL
11342 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11344 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11346 * gst/rtsp-server/rtsp-client.c:
11347 client: don't leak transports
11349 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11351 * gst/rtsp-server/rtsp-client.c:
11352 rtsp-client: free transport on no_stream in SETUP handler
11354 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11356 * gst/rtsp-server/rtsp-client.c:
11357 rtsp-client: changed session media iteration
11358 In client_unlink_session: now don't iterate in session->medias
11359 list where items are removed by gst_rtsp_session_release_media.
11360 Instead, repeatedly remove the first item.
11362 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11364 * gst/rtsp-server/rtsp-client.c:
11365 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11366 GstRTSPSessionMedia is not a GObject type. When the
11367 GstRTSPSession is freed, it will free the media.
11369 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11371 * gst/rtsp-server/rtsp-media-factory.c:
11372 factory: plug pad leak in collect_streams
11373 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11374 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11375 will take one reference, and the other reference will otherwise
11376 give a memory leak.
11378 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11381 configure: suppress some warnings when debug is disabled
11382 Warnings about unused variables should be suppressed if core has the
11383 debug system disabled.
11384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11386 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11388 * docs/libs/Makefile.am:
11389 docs: fix build in uninstalled setup
11390 Include gst-plugins-base libs properly.
11392 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11394 * docs/libs/gst-rtsp-server.types:
11395 docs: include headers defining rtsp-server object types
11396 Fixes compiler warnings during docs build.
11397 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11399 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11402 configure: Add warning flags for compiler when configuring
11403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11405 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11408 Automatic update of common submodule
11409 From 03a0e57 to 98e386f
11411 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11414 Automatic update of common submodule
11415 From 1fab359 to 03a0e57
11417 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11419 * gst/rtsp-server/rtsp-client.c:
11420 client: fix GSocketAddress leak in gst_rtsp_client_accept
11421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11423 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11426 Automatic update of common submodule
11427 From f1b5a96 to 1fab359
11429 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11432 Automatic update of common submodule
11433 From 92b7266 to f1b5a96
11435 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11438 Automatic update of common submodule
11439 From ec1c4a8 to 92b7266
11441 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11444 Automatic update of common submodule
11445 From 3429ba6 to ec1c4a8
11447 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11449 * gst/rtsp-server/rtsp-auth.c:
11450 * gst/rtsp-server/rtsp-client.c:
11451 * gst/rtsp-server/rtsp-media-factory-uri.c:
11452 * gst/rtsp-server/rtsp-server.c:
11453 rtsp: fix compiler warnings
11454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11456 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11459 Automatic update of common submodule
11460 From dc70203 to 3429ba6
11462 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11464 * gst/rtsp-server/rtsp-client.c:
11465 * gst/rtsp-server/rtsp-media-factory.c:
11466 * gst/rtsp-server/rtsp-media-factory.h:
11467 * gst/rtsp-server/rtsp-media.c:
11468 * gst/rtsp-server/rtsp-media.h:
11469 * gst/rtsp-server/rtsp-server.c:
11470 * gst/rtsp-server/rtsp-server.h:
11471 * gst/rtsp-server/rtsp-session-pool.c:
11472 * gst/rtsp-server/rtsp-session-pool.h:
11473 rtsp-server: port to new thread API
11475 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11478 Automatic update of common submodule
11479 From 6db25be to dc70203
11481 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11483 * gst/rtsp-server/rtsp-auth.c:
11484 * gst/rtsp-server/rtsp-auth.h:
11485 * gst/rtsp-server/rtsp-client.c:
11486 rtsp-server: Fix compilation and compiler warnings
11488 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11492 * gst/rtsp-server/Makefile.am:
11493 configure: Modernize autotools setup a bit
11494 Also we now only create tar.bz2 and tar.xz tarballs.
11496 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11499 Automatic update of common submodule
11500 From 464fe15 to 6db25be
11502 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11505 Automatic update of common submodule
11506 From 7fda524 to 464fe15
11508 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11511 * docs/libs/Makefile.am:
11512 * docs/version.entities.in:
11513 * gst-rtsp.spec.in:
11514 * gst/rtsp-server/Makefile.am:
11515 * pkgconfig/Makefile.am:
11516 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11517 * pkgconfig/gstreamer-rtsp-server.pc.in:
11518 * tests/Makefile.am:
11519 rtsp-server: Update versioning
11521 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11523 Merge remote-tracking branch 'origin/0.10'
11525 gst/rtsp-server/rtsp-session-pool.c
11527 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11529 * gst/rtsp-server/rtsp-session-pool.c:
11530 rtsp-server: Don't use deprecated GLib API
11532 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11534 Replace master with 0.11
11536 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11538 Merge branch 'master' into 0.11
11540 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11542 Merge branch 'master' into 0.11
11544 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11547 A couple minor typo fixes
11549 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11551 * gst/rtsp-server/rtsp-media.c:
11552 media: fix state of the appqueue
11554 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11556 * gst/rtsp-server/rtsp-media-factory-uri.c:
11557 factory: use videoconvert
11559 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11561 * gst/rtsp-server/rtsp-media-factory-uri.c:
11562 factory: change to new style caps
11564 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11566 * gst/rtsp-server/rtsp-client.c:
11567 * gst/rtsp-server/rtsp-client.h:
11568 * gst/rtsp-server/rtsp-media-factory-uri.c:
11569 * gst/rtsp-server/rtsp-media.c:
11570 * gst/rtsp-server/rtsp-server.c:
11571 * gst/rtsp-server/rtsp-server.h:
11572 * gst/rtsp-server/rtsp-session-pool.c:
11573 rtsp-server: port to GIO
11576 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11579 configure: fix build
11581 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11584 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11585 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11587 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11590 * examples/Makefile.am:
11591 First rule of gst-rtsp-server club: don't talk about gst-phonon
11593 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11596 * pkgconfig/Makefile.am:
11597 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11598 * pkgconfig/gstreamer-rtsp-server.pc.in:
11599 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11600 For consistency with all other modules.
11602 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11604 * gst/rtsp-server/rtsp-client.c:
11605 rtsp-client: update for new map API
11607 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11610 * bindings/Makefile.am:
11611 * bindings/python/Makefile.am:
11612 * bindings/python/arg-types.py:
11613 * bindings/python/codegen/Makefile.am:
11614 * bindings/python/codegen/__init__.py:
11615 * bindings/python/codegen/argtypes.py:
11616 * bindings/python/codegen/code-coverage.py:
11617 * bindings/python/codegen/codegen.py:
11618 * bindings/python/codegen/definitions.py:
11619 * bindings/python/codegen/defsparser.py:
11620 * bindings/python/codegen/docextract.py:
11621 * bindings/python/codegen/docgen.py:
11622 * bindings/python/codegen/fileprefix.override:
11623 * bindings/python/codegen/fileprefixmodule.c:
11624 * bindings/python/codegen/h2def.py:
11625 * bindings/python/codegen/mergedefs.py:
11626 * bindings/python/codegen/mkskel.py:
11627 * bindings/python/codegen/override.py:
11628 * bindings/python/codegen/reversewrapper.py:
11629 * bindings/python/codegen/scmexpr.py:
11630 * bindings/python/rtspserver-types.defs:
11631 * bindings/python/rtspserver.defs:
11632 * bindings/python/rtspserver.override:
11633 * bindings/python/rtspservermodule.c:
11634 * bindings/python/test.py:
11636 python: remove pygst-based python bindings
11637 pygi is the future, apparently.
11639 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
11642 Automatic update of common submodule
11643 From c463bc0 to 7fda524
11645 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11648 Automatic update of common submodule
11649 From 2a59016 to c463bc0
11651 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11654 Automatic update of common submodule
11655 From 0807187 to 2a59016
11657 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11660 Automatic update of common submodule
11661 From 11f0cd5 to 0807187
11663 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11665 * examples/test-auth.c:
11666 example: update for new caps
11668 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11670 * examples/test-video.c:
11671 * gst/rtsp-server/rtsp-client.c:
11672 * gst/rtsp-server/rtsp-media-factory-uri.c:
11673 * gst/rtsp-server/rtsp-media.c:
11674 * gst/rtsp-server/rtsp-media.h:
11675 * gst/rtsp-server/rtsp-session.c:
11676 * gst/rtsp-server/rtsp-session.h:
11677 rtsp-server: port some more to 0.11
11679 Remove bufferlist stuff
11680 Update for new API.
11681 Add queue before appsink now that preroll-queue-len is gone.
11682 Update for request pad changes.
11684 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11686 Merge branch 'master' into 0.11
11688 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11690 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11691 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11692 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11694 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11696 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11697 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11698 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11700 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11702 Merge branch 'master' into 0.11
11704 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11706 * gst/rtsp-server/rtsp-media.c:
11707 * gst/rtsp-server/rtsp-media.h:
11708 media: add a seekable boolean
11709 Maintain the seekable state with a new variable instead of reusing the
11712 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
11714 * gst/rtsp-server/rtsp-media.c:
11715 Disallow seek in live media
11717 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11719 Merge branch 'master' into 0.11
11721 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
11723 * gst/rtsp-server/rtsp-server.c:
11724 #ifdef statements for windows socket creation were missing
11726 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
11729 Automatic update of common submodule
11730 From a39eb83 to 11f0cd5
11732 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
11735 Automatic update of common submodule
11736 From 605cd9a to a39eb83
11738 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11740 Merge branch 'master' into 0.11
11742 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11744 * gst/rtsp-server/rtsp-client.c:
11745 client: use method to access property
11747 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11749 * gst/rtsp-server/rtsp-media-factory.c:
11750 * gst/rtsp-server/rtsp-media-factory.h:
11751 media-factory: add protocols property
11752 Add a property to configure the allowed protocols in the media created from the
11755 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11757 * gst/rtsp-server/rtsp-media-factory.c:
11758 * gst/rtsp-server/rtsp-media-factory.h:
11759 media-factory: add media-configure signal
11760 Add signal to allow the application to configure the media after it was created
11763 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11765 * gst/rtsp-server/rtsp-client.c:
11766 client: use method to access property
11768 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11770 * gst/rtsp-server/rtsp-media-factory.c:
11771 * gst/rtsp-server/rtsp-media-factory.h:
11772 media-factory: add protocols property
11773 Add a property to configure the allowed protocols in the media created from the
11776 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11778 * gst/rtsp-server/rtsp-media-factory.c:
11779 * gst/rtsp-server/rtsp-media-factory.h:
11780 media-factory: add media-configure signal
11781 Add signal to allow the application to configure the media after it was created
11784 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11786 Merge branch 'master' into 0.11
11788 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11790 * gst/rtsp-server/rtsp-client.c:
11791 client: use media multicast group
11793 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11795 * gst/rtsp-server/rtsp-media-factory.h:
11796 * gst/rtsp-server/rtsp-server.h:
11797 * gst/rtsp-server/rtsp-session-pool.h:
11798 * gst/rtsp-server/rtsp-session.h:
11801 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11803 * gst/rtsp-server/rtsp-client.c:
11804 * gst/rtsp-server/rtsp-sdp.h:
11805 sdp: copy and free the server ip address
11806 Copy and free the server ip address to make memory management easier later.
11808 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11810 * gst/rtsp-server/rtsp-media-factory.c:
11811 media-factory: configure multicast in media
11813 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11815 * gst/rtsp-server/rtsp-media.c:
11816 * gst/rtsp-server/rtsp-media.h:
11817 media: add property for multicast group
11818 Add a property to configure the multicast group in the media.
11819 Based on patches from Marc Leeman and Robert Krakora.
11821 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11823 * gst/rtsp-server/rtsp-media-factory.c:
11824 * gst/rtsp-server/rtsp-media-factory.h:
11825 media-factory: add property for multicast group
11826 Add a property to configure the multicast group in the media factory.
11827 Based on patches from Marc Leeman and Robert Krakora.
11829 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11831 * gst/rtsp-server/rtsp-client.c:
11832 client: do configuration of transport in one place
11833 Move the configuration of the transport destination address to where we also
11834 configure the other bits.
11836 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11838 * gst/rtsp-server/rtsp-client.c:
11839 client: use media multicast group
11841 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11843 * gst/rtsp-server/rtsp-media-factory.h:
11844 * gst/rtsp-server/rtsp-server.h:
11845 * gst/rtsp-server/rtsp-session-pool.h:
11846 * gst/rtsp-server/rtsp-session.h:
11849 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11851 * gst/rtsp-server/rtsp-client.c:
11852 * gst/rtsp-server/rtsp-sdp.h:
11853 sdp: copy and free the server ip address
11854 Copy and free the server ip address to make memory management easier later.
11856 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11858 * gst/rtsp-server/rtsp-media-factory.c:
11859 media-factory: configure multicast in media
11861 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11863 * gst/rtsp-server/rtsp-media.c:
11864 * gst/rtsp-server/rtsp-media.h:
11865 media: add property for multicast group
11866 Add a property to configure the multicast group in the media.
11867 Based on patches from Marc Leeman and Robert Krakora.
11869 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11871 * gst/rtsp-server/rtsp-media-factory.c:
11872 * gst/rtsp-server/rtsp-media-factory.h:
11873 media-factory: add property for multicast group
11874 Add a property to configure the multicast group in the media factory.
11875 Based on patches from Marc Leeman and Robert Krakora.
11877 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11879 * gst/rtsp-server/rtsp-client.c:
11880 client: do configuration of transport in one place
11881 Move the configuration of the transport destination address to where we also
11882 configure the other bits.
11884 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11886 Merge branch 'master' into 0.11
11888 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11890 * gst/rtsp-server/rtsp-client.c:
11891 client: destroy pipeline on client disconnect with no prior TEARDOWN.
11892 The problem occurs when the client abruptly closes the connection without
11893 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
11894 server is where the pipeline gets torn down. Since this handler is not called,
11895 the pipeline remains and is up and running. Subsequent clients get their own
11896 pipelines and if the do not issue TEARDOWNs then those pipelines will also
11897 remain up and running. This is a resource leak.
11899 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11901 Merge branch 'master' into 0.11
11903 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
11905 * gst/rtsp-server/rtsp-media-factory.c:
11906 * gst/rtsp-server/rtsp-media-factory.h:
11907 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
11908 For example, it can be used to retrieve source elements like appsrc, in a more
11909 convenient way than subclassing get_element.
11911 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11913 Merge branch 'master' into 0.11
11915 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
11917 * gst/rtsp-server/rtsp-server.c:
11918 rtsp-server: hold on to reference while using object
11920 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11922 * gst/rtsp-server/rtsp-media.c:
11925 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11928 configure: use unstable api
11930 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
11932 * gst/rtsp-server/rtsp-client.c:
11933 client: fix reference counting
11935 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
11937 * gst/rtsp-server/rtsp-client.c:
11938 * gst/rtsp-server/rtsp-media.c:
11939 fix compiler warnings about unused variables
11941 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
11943 * examples/test-launch.c:
11944 * examples/test-readme.c:
11945 * examples/test-uri.c:
11946 * examples/test-video.c:
11947 examples: tell rtsp uri when ready
11949 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
11952 Automatic update of common submodule
11953 From 69b981f to 605cd9a
11955 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11957 * gst/rtsp-server/rtsp-client.c:
11958 client: update for buffer API change
11960 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11962 * gst/rtsp-server/Makefile.am:
11963 Makefile.am: 0.10 => @GST_MAJORMINOR@
11965 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11967 * gst/rtsp-server/rtsp-media-factory-uri.c:
11968 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
11970 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11972 * gst/rtsp-server/.gitignore:
11973 .gitignore: 0.10 => 0.11
11975 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11977 * gst/rtsp-server/Makefile.am:
11978 Makefile.am: 0.10 => @GST_MAJORMINOR@
11980 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11982 Merge branch 'master' into 0.11
11984 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
11987 Automatic update of common submodule
11988 From 9e5bbd5 to 69b981f
11990 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
11993 Automatic update of common submodule
11994 From fd35073 to 9e5bbd5
11996 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
11999 Automatic update of common submodule
12000 From 46dfcea to fd35073
12002 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12004 * gst/rtsp-server/rtsp-media-factory-uri.c:
12005 * gst/rtsp-server/rtsp-media.c:
12006 media: port to new caps API
12008 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12010 Merge branch 'master' into 0.11
12012 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12014 * bindings/vala/gst-rtsp-server-0.10.vapi:
12015 Updated Vala bindings.
12016 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12018 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12020 * gst/rtsp-server/rtsp-server.c:
12021 * gst/rtsp-server/rtsp-server.h:
12022 Add a signal for newly connected clients.
12023 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12025 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12027 * bindings/python/rtspserver.override:
12028 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12030 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12032 * gst/rtsp-server/Makefile.am:
12033 * gst/rtsp-server/rtsp-client.c:
12034 * gst/rtsp-server/rtsp-funnel.c:
12035 * gst/rtsp-server/rtsp-funnel.h:
12036 * gst/rtsp-server/rtsp-media.c:
12037 rtsp-server: port to 0.11
12039 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12044 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12046 Merge branch 'master' into 0.11
12051 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12054 Automatic update of common submodule
12055 From c3cafe1 to 46dfcea
12057 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12059 * bindings/python/Makefile.am:
12060 * bindings/python/rtspserver.defs:
12061 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12063 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12065 * bindings/python/arg-types.py:
12066 python bindings: add GstRTSPUrlParam
12067 Needed to implement MediaFactory virtual proxies
12069 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12071 * bindings/python/arg-types.py:
12072 python bindings: fix returning GstRTSPUrl types
12074 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12076 * bindings/python/arg-types.py:
12077 python bindings: add arg type for GstRTSPUrl
12079 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12081 * bindings/python/rtspserver.defs:
12082 python bindings: fix the definition of MediaFactory.collect_stream
12084 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12087 Automatic update of common submodule
12088 From 1ccbe09 to c3cafe1
12090 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12093 Automatic update of common submodule
12094 From 193b717 to 1ccbe09
12096 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12099 Automatic update of common submodule
12100 From b77e2bf to 193b717
12102 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12105 build: Include lcov.mak to allow test coverage report generation
12107 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12110 Automatic update of common submodule
12111 From d8814b6 to b77e2bf
12113 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12116 Automatic update of common submodule
12117 From 6aaa286 to d8814b6
12119 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12122 Automatic update of common submodule
12123 From 6aec6b9 to 6aaa286
12125 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12128 autogen: wingo signed comment
12130 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12132 * gst/rtsp-server/rtsp-session-pool.c:
12133 session: use full charset for RTSP session ID
12134 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12135 session ID more difficult.
12136 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12138 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12140 * gst/rtsp-server/Makefile.am:
12141 rtsp-server: Don't install the funnel header
12143 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12146 Automatic update of common submodule
12147 From 1de7f6a to 6aec6b9
12149 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12152 configure: require core/base 0.10.31
12153 Needed at least for gst_plugin_feature_rank_compare_func().
12155 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12158 Automatic update of common submodule
12159 From f94d739 to 1de7f6a
12161 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12163 * gst/rtsp-server/rtsp-media.c:
12164 media: remove more unused code
12166 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12168 * gst/rtsp-server/rtsp-media.c:
12169 * gst/rtsp-server/rtsp-media.h:
12170 media: remove duplicate filtering
12171 Remove the duplicate filtering code now that we have a released -good version.
12172 Give a warning instead.
12174 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12176 * gst/rtsp-server/rtsp-media-factory.c:
12177 * gst/rtsp-server/rtsp-media.c:
12178 media: fix default buffer size
12180 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12182 * gst/rtsp-server/rtsp-media-factory.c:
12183 * gst/rtsp-server/rtsp-media-factory.h:
12184 media-factory: add property to configure the buffer-size
12185 Add a property to configure the kernel UDP buffer size.
12187 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12189 * gst/rtsp-server/rtsp-media.c:
12190 * gst/rtsp-server/rtsp-media.h:
12191 media: add property to configure kernel buffer sizes
12192 Add a property to configure the kernel UDP buffer size.
12194 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12197 configure: set PYGOBJECT_REQ before using it
12198 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12200 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12202 * docs/Makefile.am:
12203 docs: recursive into sub-directories on 'make upload'
12205 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12207 * docs/libs/gst-rtsp-server-docs.sgml:
12208 * docs/version.entities.in:
12209 docs: mention full version these docs are for, not just major-minor
12211 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12214 back to development
12216 === release 0.10.8 ===
12218 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12223 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12225 * gst/rtsp-server/rtsp-server.c:
12226 rtsp-server: clarify docs a little
12228 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12230 * gst/rtsp-server/rtsp-media.c:
12231 media: init debug category before starting thread
12233 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12235 * gst/rtsp-server/rtsp-auth.c:
12236 auth: add realm to make it more spec compliant
12238 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12240 * gst/rtsp-server/rtsp-server.c:
12241 * gst/rtsp-server/rtsp-server.h:
12242 server: add locking
12244 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12246 * examples/test-video.c:
12247 example: improve example docs a little
12249 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12251 * gst/rtsp-server/rtsp-server.c:
12252 server: ensure the watch has a ref to the server
12254 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12256 * gst/rtsp-server/rtsp-server.c:
12257 server: simpify channel function
12259 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12261 * gst/rtsp-server/rtsp-server.c:
12262 * gst/rtsp-server/rtsp-server.h:
12263 server: simplify management of channel and source
12264 We don't need to keep around the channel and source objects. Let the mainloop
12265 and the source manage the source and channel respectively.
12267 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12273 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12275 * tests/.gitignore:
12276 * tests/Makefile.am:
12277 * tests/test-cleanup.c:
12278 tests: add tests directory and cleanup test
12280 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12282 * gst/rtsp-server/rtsp-media-factory-uri.c:
12283 * gst/rtsp-server/rtsp-media-factory.c:
12284 * gst/rtsp-server/rtsp-media-mapping.c:
12285 * gst/rtsp-server/rtsp-media.c:
12286 * gst/rtsp-server/rtsp-session-pool.c:
12287 * gst/rtsp-server/rtsp-session.c:
12288 server: improve debugging in various objects
12290 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12292 * gst/rtsp-server/rtsp-server.c:
12293 server: chain up to the parent finalize
12295 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12297 * bindings/python/rtspserver-types.defs:
12298 * bindings/python/rtspserver.defs:
12299 * bindings/python/rtspserver.override:
12300 * bindings/python/test.py:
12301 gst-rtsp-server: update python bindings
12303 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12305 * gst/rtsp-server/rtsp-client.c:
12306 client: use the response from the clientstate
12307 Create the response object only once and store in the client state.
12308 Make all methods use the state response,
12310 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12312 * gst/rtsp-server/rtsp-server.c:
12313 server: use signal to keep track of clients
12314 Keep track of all the clients that the server creates and remove them when they
12315 fire the 'closed' signal.
12317 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12319 * gst/rtsp-server/rtsp-client.c:
12320 * gst/rtsp-server/rtsp-client.h:
12321 client: emit signal when closing
12323 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12325 * examples/.gitignore:
12326 * examples/Makefile.am:
12327 * examples/test-auth.c:
12328 * examples/test-video.c:
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-media-factory.c:
12333 * gst/rtsp-server/rtsp-media.c:
12334 * gst/rtsp-server/rtsp-media.h:
12335 * gst/rtsp-server/rtsp-session-pool.h:
12336 * gst/rtsp-server/rtsp-session.h:
12337 media: enable per factory authorisations
12338 Allow for adding a GstRTSPAuth on the factory and media level and check
12339 permissions when accessing the factory.
12340 Add hints to the auth methods for future more fine grained authorisation.
12341 Add example application for per factory authentication.
12343 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12345 * gst/rtsp-server/rtsp-auth.c:
12346 * gst/rtsp-server/rtsp-auth.h:
12347 * gst/rtsp-server/rtsp-client.c:
12348 * gst/rtsp-server/rtsp-client.h:
12349 * gst/rtsp-server/rtsp-params.c:
12350 * gst/rtsp-server/rtsp-params.h:
12351 rtsp-server: Pass ClientState structure arround
12352 Pass the collected information for the ongoing request in a GstRTSPClientState
12353 structure that we can then pass around to simplify the method arguments. This
12354 will also be handy when we implement logging functionality.
12356 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12358 * gst/rtsp-server/rtsp-media-factory.c:
12359 * gst/rtsp-server/rtsp-media-factory.h:
12360 media-factory: add methods to configure authorisation
12362 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12364 * gst/rtsp-server/rtsp-client.c:
12365 client: unref auth in finalize
12367 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12369 * gst/rtsp-server/rtsp-server.c:
12370 server: unref auth in finalize
12372 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12374 * docs/libs/gst-rtsp-server-docs.sgml:
12375 * docs/libs/gst-rtsp-server-sections.txt:
12376 * docs/libs/gst-rtsp-server.types:
12377 docs: add more docs
12379 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12381 * gst/rtsp-server/rtsp-server.c:
12382 * gst/rtsp-server/rtsp-server.h:
12383 server: separate create and accept
12384 Create separate create and accept methods so that subclasses can create custom
12386 Configure the server in the client object and prepare for keeping track of
12389 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12391 * gst/rtsp-server/rtsp-client.c:
12392 * gst/rtsp-server/rtsp-client.h:
12393 client: add support for setting the server.
12394 Add support for keeping a ref to the server that started this client
12397 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12399 * gst/rtsp-server/rtsp-auth.c:
12400 auth: fix memleak and add some docs
12401 Fix a memleak of the basic auth token.
12402 Add docs for the helper function
12404 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12406 * gst/rtsp-server/rtsp-auth.c:
12407 * gst/rtsp-server/rtsp-auth.h:
12408 * gst/rtsp-server/rtsp-client.c:
12409 client: delegate setup of auth to the manager
12410 Delegate the configuration of the authentication tokens to the manager object
12413 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12415 * examples/test-video.c:
12416 * gst/rtsp-server/Makefile.am:
12417 * gst/rtsp-server/rtsp-auth.c:
12418 * gst/rtsp-server/rtsp-auth.h:
12419 * gst/rtsp-server/rtsp-client.c:
12420 * gst/rtsp-server/rtsp-client.h:
12421 * gst/rtsp-server/rtsp-server.c:
12422 * gst/rtsp-server/rtsp-server.h:
12423 auth: add authentication object
12424 Add an object that can check the authorization of requests.
12425 Implement basic authentication.
12426 Add example authentication to test-video
12428 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12430 * gst/rtsp-server/rtsp-server.c:
12431 * gst/rtsp-server/rtsp-server.h:
12432 server: move includes back
12433 the includes are needed for sockaddr_in.
12435 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12437 * gst/rtsp-server/rtsp-client.c:
12438 * gst/rtsp-server/rtsp-client.h:
12439 * gst/rtsp-server/rtsp-server.c:
12440 * gst/rtsp-server/rtsp-server.h:
12441 rtsp: move network includes where they are needed
12443 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12445 * gst/rtsp-server/rtsp-media.h:
12446 rtsp-media.h: Minor corrections in comments.
12449 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12452 Automatic update of common submodule
12453 From e572c87 to f94d739
12455 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12459 * docs/libs/.gitignore:
12460 * examples/.gitignore:
12461 * gst/rtsp-server/.gitignore:
12464 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12466 * docs/libs/Makefile.am:
12467 docs: We don't build ps/pdf for API reference docs
12469 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12472 Automatic update of common submodule
12473 From ccbaa85 to e572c87
12475 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12478 Automatic update of common submodule
12479 From 46445ad to ccbaa85
12481 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12483 * gst/rtsp-server/Makefile.am:
12484 * gst/rtsp-server/rtsp-funnel.c:
12485 * gst/rtsp-server/rtsp-funnel.h:
12486 * gst/rtsp-server/rtsp-media.c:
12487 funnel: rename fsfunnel to rtspfunnel
12488 Rename the funnel to avoid conflicts with the farsight one.
12490 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12492 * gst/rtsp-server/Makefile.am:
12493 * gst/rtsp-server/fs-funnel.c:
12494 * gst/rtsp-server/fs-funnel.h:
12495 * gst/rtsp-server/rtsp-media.c:
12496 rtsp-media: add and use fsfunnel
12497 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12498 select-all property that we need.
12500 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12502 * gst/rtsp-server/Makefile.am:
12503 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12504 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12505 for the g-ir-compiler, rather than just assuming the env var has
12508 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12515 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12517 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12520 * gst/rtsp-server/Makefile.am:
12521 gobject-introspection: fix g-i build for uninstalled setup
12522 Requires gst-plugins-base git (> 0.10.31.2).
12524 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12526 * examples/test-uri.c:
12527 examples: add some more options and comments
12529 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12531 * gst/rtsp-server/rtsp-media-factory-uri.c:
12532 factory-uri: use right property type
12534 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12536 * gst/rtsp-server/rtsp-media-factory-uri.c:
12537 factory-uri: attempt to configure buffer-lists
12538 Attempt to configure buffer lists in the payloader for improved performance.
12540 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12542 * gst/rtsp-server/rtsp-media.c:
12543 media: attempt to configure bigger UDP buffers
12544 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12545 send buffers with high bitrate streams.
12547 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12549 * gst/rtsp-server/rtsp-client.c:
12550 client: use the socket length from getsockname
12551 Use the length returned by getsockname to perform the getnameinfo call because
12552 the size can depend on the socket type and platform.
12555 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12557 * docs/libs/gst-rtsp-server-docs.sgml:
12558 * docs/libs/gst-rtsp-server-sections.txt:
12559 docs: add uri factory to the docs
12561 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12563 * gst/rtsp-server/rtsp-client.c:
12564 * gst/rtsp-server/rtsp-media.h:
12567 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12569 * gst/rtsp-server/rtsp-client.c:
12570 * gst/rtsp-server/rtsp-media.c:
12571 * gst/rtsp-server/rtsp-media.h:
12572 * gst/rtsp-server/rtsp-session.c:
12573 * gst/rtsp-server/rtsp-session.h:
12574 rtsp-server: add support for buffer lists
12575 Add support for sending bufferlists received from appsink.
12578 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12580 * gst/rtsp-server/rtsp-client.c:
12581 * gst/rtsp-server/rtsp-media.c:
12582 * gst/rtsp-server/rtsp-media.h:
12583 * gst/rtsp-server/rtsp-sdp.c:
12584 media: make method to retrieve the play range
12585 Make a method to retrieve the playback range so that we can conditionally create
12586 a different range for the SDP and the PLAY requests.
12588 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12590 * gst/rtsp-server/rtsp-media.c:
12591 * gst/rtsp-server/rtsp-media.h:
12592 media: add signal to notify of state changes
12594 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12596 * gst/rtsp-server/rtsp-client.h:
12597 client: cleanup headers
12599 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12601 * gst/rtsp-server/rtsp-client.c:
12604 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12606 * gst/rtsp-server/rtsp-media-factory-uri.c:
12607 * gst/rtsp-server/rtsp-media-factory-uri.h:
12608 factory-uri: add support for gstpay
12609 Add an option to prefer gstpay over decoder + raw payloader.
12611 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12613 * gst/rtsp-server/rtsp-media-factory-uri.c:
12614 * gst/rtsp-server/rtsp-media-factory-uri.h:
12615 factory-uri: rework the autoplugger.
12616 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12619 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12621 * gst/rtsp-server/rtsp-media-factory-uri.c:
12622 factory-uri: use better factory filter
12623 Make better payloader filter based on autoplug rank and RTP use case.
12625 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12628 Automatic update of common submodule
12629 From 169462a to 46445ad
12631 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12633 * gst/rtsp-server/rtsp-server.c:
12634 server: set SO_REUSEADDR before bind
12635 Set the SO_REUSEADDR _before_ bind() to make it actually work.
12637 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12639 * gst/rtsp-server/rtsp-media.c:
12640 * gst/rtsp-server/rtsp-media.h:
12641 media: emit prepared signal when prepared
12642 Make a 'prepared' signal and emit it when we successfully prepared the element.
12643 This signal can be used to configure the media object after it has been prepared
12646 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
12649 Automatic update of common submodule
12650 From 011bcc8 to 169462a
12652 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
12654 python an optional dependency
12655 * configure.ac: Move up valgrind and g-i checks. Make the python
12656 dependency optional, as it was before.
12658 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12660 Merge branch 'master' into 0.11
12665 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12667 * gst/rtsp-server/rtsp-media.c:
12668 media: update range when active clients changed
12669 When we changed the number of active clients, update the current range
12670 information because we want the second client connecting to a shared resource
12671 continue from where the stream currently.
12673 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12675 * gst/rtsp-server/rtsp-media-factory-uri.c:
12676 * gst/rtsp-server/rtsp-media-factory-uri.h:
12677 factory-uri: add colorspace and fix pt
12678 Rework the way we pass data to the autoplugger.
12679 When we have raw caps, plug a converter element to make pluggin to raw
12680 payloaders more successful.
12681 Make sure all dynamically plugged payloaders have a unique payload types.
12683 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12685 * examples/Makefile.am:
12686 * examples/test-uri.c:
12687 example: add example of the uri factory
12689 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12691 * gst/rtsp-server/Makefile.am:
12692 * gst/rtsp-server/rtsp-media-factory-uri.c:
12693 * gst/rtsp-server/rtsp-media-factory-uri.h:
12694 * gst/rtsp-server/rtsp-server.h:
12695 factory-uri: add a factory to stream any URI
12696 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
12699 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12701 * gst/rtsp-server/rtsp-media.c:
12702 * gst/rtsp-server/rtsp-media.h:
12703 media: ignore spurious ASYNC_DONE messages
12704 When we are dynamically adding pads, the addition of the udpsrc elements will
12705 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
12706 the real ASYNC_DONE when everything is prerolled.
12708 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12710 * gst/rtsp-server/rtsp-media-factory.c:
12711 * gst/rtsp-server/rtsp-media-factory.h:
12712 media-factory: make lock macro
12714 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
12716 * gst/rtsp-server/rtsp-client.c:
12717 rtsp-server: Remove unused variable and dead assignment
12719 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
12721 * examples/test-launch.c:
12722 * examples/test-mp4.c:
12723 * examples/test-ogg.c:
12724 * examples/test-readme.c:
12725 * examples/test-sdp.c:
12726 * examples/test-video.c:
12727 examples: Run gst-indent
12729 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
12731 * gst/rtsp-server/rtsp-client.c:
12732 * gst/rtsp-server/rtsp-media-factory.c:
12733 * gst/rtsp-server/rtsp-media-mapping.c:
12734 * gst/rtsp-server/rtsp-media.c:
12735 * gst/rtsp-server/rtsp-params.c:
12736 * gst/rtsp-server/rtsp-sdp.c:
12737 * gst/rtsp-server/rtsp-server.c:
12738 * gst/rtsp-server/rtsp-session-pool.c:
12739 * gst/rtsp-server/rtsp-session.c:
12740 rtsp-server: Run gst-indent
12741 Since it wasn't using the upstream common previously, there was no
12742 indentation check before commiting.
12744 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
12746 * gst/rtsp-server/rtsp-media-mapping.h:
12747 * gst/rtsp-server/rtsp-media.c:
12748 * gst/rtsp-server/rtsp-media.h:
12749 * gst/rtsp-server/rtsp-sdp.c:
12750 * gst/rtsp-server/rtsp-session-pool.h:
12751 * gst/rtsp-server/rtsp-session.c:
12752 * gst/rtsp-server/rtsp-session.h:
12753 rtsp-server: Some more doc fixups
12755 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12758 Makefile: Add cruft-cleaning support
12760 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12764 * docs/Makefile.am:
12765 * docs/libs/Makefile.am:
12766 * docs/libs/gst-rtsp-server-docs.sgml:
12767 * docs/libs/gst-rtsp-server-sections.txt:
12768 * docs/libs/gst-rtsp-server.types:
12769 * docs/version.entities.in:
12770 docs: Add gtk-doc build system
12772 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12774 * gst/rtsp-server/Makefile.am:
12775 Makefile.am: Use standard GIR make behaviour
12777 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12781 autogen/configure: Bring more in sync to standard gst module behaviour
12783 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12785 * gst/rtsp-server/rtsp-media.c:
12786 media: warn and fail when gstrtpbin is not found
12788 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12791 configure: open 0.11 branch
12793 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
12797 Add common submodule
12799 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
12801 * common/ChangeLog:
12802 * common/Makefile.am:
12803 * common/c-to-xml.py:
12804 * common/check.mak:
12805 * common/coverage/coverage-report-entry.pl:
12806 * common/coverage/coverage-report.pl:
12807 * common/coverage/coverage-report.xsl:
12808 * common/coverage/lcov.mak:
12809 * common/gettext.patch:
12810 * common/glib-gen.mak:
12811 * common/gst-autogen.sh:
12812 * common/gst-xmlinspect.py:
12814 * common/gstdoc-scangobj:
12815 * common/gtk-doc-plugins.mak:
12816 * common/gtk-doc.mak:
12817 * common/m4/.gitignore:
12818 * common/m4/Makefile.am:
12819 * common/m4/README:
12820 * common/m4/as-ac-expand.m4:
12821 * common/m4/as-auto-alt.m4:
12822 * common/m4/as-compiler-flag.m4:
12823 * common/m4/as-compiler.m4:
12824 * common/m4/as-docbook.m4:
12825 * common/m4/as-libtool-tags.m4:
12826 * common/m4/as-libtool.m4:
12827 * common/m4/as-python.m4:
12828 * common/m4/as-scrub-include.m4:
12829 * common/m4/as-version.m4:
12830 * common/m4/ax_create_stdint_h.m4:
12831 * common/m4/check.m4:
12832 * common/m4/glib-gettext.m4:
12833 * common/m4/gst-arch.m4:
12834 * common/m4/gst-args.m4:
12835 * common/m4/gst-check.m4:
12836 * common/m4/gst-debuginfo.m4:
12837 * common/m4/gst-default.m4:
12838 * common/m4/gst-doc.m4:
12839 * common/m4/gst-error.m4:
12840 * common/m4/gst-feature.m4:
12841 * common/m4/gst-function.m4:
12842 * common/m4/gst-gettext.m4:
12843 * common/m4/gst-glib2.m4:
12844 * common/m4/gst-libxml2.m4:
12845 * common/m4/gst-plugindir.m4:
12846 * common/m4/gst-valgrind.m4:
12847 * common/m4/gtk-doc.m4:
12848 * common/m4/introspection.m4:
12849 * common/m4/pkg.m4:
12850 * common/mangle-tmpl.py:
12851 * common/plugins.xsl:
12853 * common/release.mak:
12854 * common/scangobj-merge.py:
12855 * common/upload.mak:
12856 common: Remove static version
12858 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
12860 * common/m4/introspection.m4:
12861 Update introspection.m4 to match usage
12863 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12867 Remove old stuff from the README
12869 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12872 back to development
12874 === release 0.10.7 ===
12876 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12881 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12883 * examples/test-ogg.c:
12884 test-ogg: remove parsers
12885 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
12886 buffers with timestamps. Using the parsers also seems to break things.
12888 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12890 * bindings/vala/gst-rtsp-server-0.10.vapi:
12891 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12892 Updated Vala bindings
12894 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12896 * common/m4/introspection.m4:
12898 * gst/rtsp-server/Makefile.am:
12899 Added initial gobject-introspection support
12901 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12903 * gst/rtsp-server/rtsp-media-factory.c:
12904 media-factory: don't use host for shared hash key
12905 When we generate the key to share made between connections, don't include the
12906 host used to connect so that we can share media even if between clients that
12907 connected with localhost and ones with the ip address.
12909 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12911 * bindings/vala/Makefile.am:
12912 build: fix distcheck
12914 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12916 * bindings/vala/gst-rtsp-server-0.10.vapi:
12917 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12918 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12919 Update Vala bindings
12921 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12923 * bindings/vala/Makefile.am:
12925 Fix configure checks and installation location for Vala bindings
12928 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12931 back to development
12933 === release 0.10.6 ===
12935 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12938 configure: release 0.10.6
12940 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12942 * gst/rtsp-server/rtsp-media.c:
12943 media: help the compiler a little
12945 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12947 * gst/rtsp-server/rtsp-media.c:
12948 * gst/rtsp-server/rtsp-media.h:
12949 * gst/rtsp-server/rtsp-session.c:
12950 media: cleanup media transport before freeing
12951 Cleanup the media transport data before freeing. In particular, remove the qdata
12952 from the rtpsource object.
12954 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12956 * gst/rtsp-server/rtsp-media-factory.c:
12957 * gst/rtsp-server/rtsp-media-factory.h:
12958 * gst/rtsp-server/rtsp-media.c:
12959 * gst/rtsp-server/rtsp-media.h:
12960 media-factory: add eos-shutdown property
12961 Add an eos-shutdown property that will send an EOS to the pipeline before
12962 shutting it down. This allows for nice cleanup in case of a muxer.
12965 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12967 * gst/rtsp-server/rtsp-media.c:
12968 * gst/rtsp-server/rtsp-media.h:
12969 media: use multiudpsink send-duplicates when we can
12970 If we have a new enough multiudpsink with the send-duplicates property, use this
12971 instead of doing our own filtering. Our custom filtering code should eventually
12972 be removed when we can depend on a released -good.
12974 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12976 * gst/rtsp-server/rtsp-media.c:
12977 media: don't leak destinations
12978 Refactor and cleanup the destinations array when the stream is destroyed.
12980 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12982 * gst/rtsp-server/rtsp-media.c:
12983 * gst/rtsp-server/rtsp-media.h:
12984 media: don't add udp addresses multiple times
12985 Keep track of the udp addresses we added to udpsink and never add the same udp
12986 destination twice. This avoids duplicate packets when using multicast.
12988 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12990 * gst/rtsp-server/rtsp-server.c:
12991 server: disable use of SO_LINGER
12992 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
12993 server close()s the connection.
12995 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12997 * gst/rtsp-server/rtsp-server.c:
12998 server: use 5 second linger period in SO_LINGER
12999 Wait 5 seconds before clearing the send buffers and reseting the connection with
13000 the client when we do a close. This should be enough time to get the message to
13004 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13006 * gst/rtsp-server/rtsp-server.c:
13007 server: use SO_LINGER
13008 SO_LINGER on the socket will make sure that any pending data on the socket is
13009 flushed ASAP and that the socket connection is reset. This makes sure that the
13010 socket can be reused immediately.
13013 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13016 README: add blurb about shared media factories
13018 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13020 * gst/rtsp-server/rtsp-media.c:
13021 Add stdlib.h for atoi()
13023 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13025 * bindings/python/Makefile.am:
13026 * bindings/vala/Makefile.am:
13027 build: distcheck fixes
13028 Fix 'make distcheck', somewhat (it still fails because it tries to
13029 install files into /usr/share/vala/vapi/ irrespective of the
13030 configured prefix).
13032 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13035 configure: bump core/base requirements to released version
13036 Makes things less confusing for people.
13038 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13041 configure: fail if GStreamer core/base requirements are not met
13043 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13045 * gst/rtsp-server/rtsp-client.c:
13046 client: improve client cleanups
13047 Make sure the session does not timeout when using TCP. We need to do this
13048 because quicktime player does not send RTCP for some reason in tunneled
13050 Refactor some cleanup code.
13053 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13055 * gst/rtsp-server/rtsp-session.c:
13056 * gst/rtsp-server/rtsp-session.h:
13057 session: add support for prevent session timeouts
13058 Add an atomix counter to prevent session timeouts when we are, for example,
13059 streaming over TCP.
13061 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13063 * gst/rtsp-server/rtsp-client.c:
13064 client: fix unlink on session timeouts
13065 When our session times out, make sure we unlink all streams in this
13067 Remove the tunnelid when closing the connection.
13069 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13071 * gst/rtsp-server/rtsp-session.c:
13072 session: small cleanups
13074 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13076 * gst/rtsp-server/rtsp-client.c:
13077 client: handle lost_tunnel callbacks
13078 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13079 hashtable so that we can reuse it for when the client reopens the POST
13081 Close the connection after a TEARDOWN.
13082 Make sure or watchid is cleared when the watch is removed.
13085 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13087 * gst/rtsp-server/rtsp-client.c:
13088 * gst/rtsp-server/rtsp-media.c:
13089 * gst/rtsp-server/rtsp-sdp.c:
13090 rtsp-server: add more support for multicast
13092 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13095 * gst/rtsp-server/rtsp-media.c:
13096 * gst/rtsp-server/rtsp-media.h:
13097 media: allow configuration of allowed lower transport
13099 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13101 * gst/rtsp-server/rtsp-client.h:
13102 * gst/rtsp-server/rtsp-media.c:
13103 * gst/rtsp-server/rtsp-media.h:
13104 * gst/rtsp-server/rtsp-sdp.c:
13105 * gst/rtsp-server/rtsp-sdp.h:
13106 * gst/rtsp-server/rtsp-server.c:
13107 rtsp: keep track of server ip and ipv6
13108 Keep track of how the client connected to the server and setup the udp ports
13109 with the same protocol.
13110 Copy the server ip address in the SDP so that clients can send RTCP back to
13113 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13115 * gst/rtsp-server/rtsp-session.c:
13118 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13120 * gst/rtsp-server/rtsp-client.c:
13121 client: use right size for malloc
13123 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13125 * gst/rtsp-server/rtsp-server.c:
13126 server: comment ipv6 server listening address
13128 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13130 * gst/rtsp-server/rtsp-media.c:
13131 media: allow for ipv6 sockets
13133 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13135 * gst/rtsp-server/rtsp-server.c:
13136 * gst/rtsp-server/rtsp-server.h:
13137 server: rework server part
13138 Allow setting a bind address, make sure we can deal with ipv6.
13139 Remove the port property and change with the service property.
13141 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13143 * gst/rtsp-server/rtsp-media.h:
13144 media: update comments a little
13146 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13148 * gst/rtsp-server/rtsp-client.c:
13149 client: make content-base better
13150 Use the URI formatting functions to make a content-base. Also make sure that
13151 there is a trailing / at the end.
13153 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13155 * gst/rtsp-server/rtsp-client.c:
13156 client: guard against invalid paths
13158 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13160 * examples/test-video.c:
13161 test: catch server bind errors
13163 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13165 * gst/rtsp-server/rtsp-media.c:
13166 rtspmedia: emit "unprepared" if _prepare fails.
13167 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13168 media object is removed from its factory's cache.
13170 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13172 * gst/rtsp-server/rtsp-media.c:
13173 media: collect media position when seek completes
13175 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13177 * gst/rtsp-server/rtsp-client.c:
13178 client: call unlink_streams in client finalize
13181 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13183 * gst/rtsp-server/rtsp-media.c:
13184 media: limit the time to wait to something huge
13185 Avoid waiting forever but limit the timeout to 20 seconds.
13187 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13189 * gst/rtsp-server/rtsp-sdp.c:
13190 sdp: reindent and check for prepared status
13192 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13194 * gst/rtsp-server/rtsp-media.c:
13195 * gst/rtsp-server/rtsp-media.h:
13196 * gst/rtsp-server/rtsp-session.c:
13197 media: avoid doing _get_state() for state changes
13198 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13199 until the media is prerolled or in error. This avoids doing a blocking call of
13200 gst_element_get_state() that can cause lockups when there is an error.
13203 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13205 * gst/rtsp-server/rtsp-media.c:
13208 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13210 * gst/rtsp-server/rtsp-media-factory.c:
13211 media-factory: better error handling
13212 Improve the error handling a bit.
13214 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13216 * gst/rtsp-server/rtsp-client.c:
13217 client: rework transport parsing
13218 Rework the transport parsing code so that we can ignore transports we don't
13219 support instead of just picking the first one we can parse.
13220 Configure a (for now hardcoded) destination for multicast transports.
13222 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13224 * gst/rtsp-server/rtsp-media.c:
13225 media: set multicast sink parameters
13226 Disable loop and automatic multicast join on the udpsink elements.
13227 Add some more debug info.
13228 Reset some state variables in the right place.
13229 Use the right port numbers for multicast.
13231 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13233 * gst/rtsp-server/rtsp-session.c:
13234 session: handle transport setup correctly
13235 Handle UDP, MCAST and TCP transport negotiation more correctly.
13236 Store the server session SSRC in the transport.
13238 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13240 * gst/rtsp-server/rtsp-client.c:
13241 rtsp-client: implement error_full
13242 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13245 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13248 * gst/rtsp-server/rtsp-client.c:
13249 * gst/rtsp-server/rtsp-server.c:
13250 docs: update docs and comments
13252 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13254 * gst/rtsp-server/rtsp-sdp.c:
13255 sdp: make server work better when behind a proxy
13257 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13259 * gst/rtsp-server/rtsp-client.c:
13260 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13262 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13264 * gst/rtsp-server/rtsp-client.c:
13265 * gst/rtsp-server/rtsp-media-factory.c:
13266 * gst/rtsp-server/rtsp-media-mapping.c:
13267 * gst/rtsp-server/rtsp-media.c:
13268 * gst/rtsp-server/rtsp-server.c:
13269 * gst/rtsp-server/rtsp-session-pool.c:
13270 * gst/rtsp-server/rtsp-session.c:
13271 Use GStreamer's debugging subsystem
13273 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13275 * gst/rtsp-server/rtsp-media-factory.c:
13276 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13278 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13281 back to development
13283 === release 0.10.5 ===
13285 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13290 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13293 configure: bump required versions
13295 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13297 * gst/rtsp-server/rtsp-client.c:
13298 client: call weak-unref on client->sessions from finalize
13301 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13303 * gst/rtsp-server/rtsp-media.c:
13304 media: Fixed crasher where caps got unref'ed too often
13306 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13309 * pkgconfig/.gitignore:
13310 * pkgconfig/Makefile.am:
13311 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13312 Added pkg-config file to use gst-rtsp-server uninstalled
13314 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13316 * gst/rtsp-server/rtsp-media.c:
13317 media: add some docs
13319 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13321 * gst/rtsp-server/rtsp-client.c:
13322 rtsp: Use gst_rtsp_watch_send_message().
13323 Use gst_rtsp_watch_send_message() since the old API which used
13324 gst_rtsp_watch_queue_message() has been deprecated.
13326 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13329 back to development
13331 === release 0.10.4 ===
13333 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13338 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13340 * gst/rtsp-server/rtsp-client.c:
13341 * gst/rtsp-server/rtsp-session.c:
13342 * gst/rtsp-server/rtsp-session.h:
13343 rtsp: allocate channels in TCP mode
13344 When the client does not provide us with channels in TCP mode, allocate channels
13347 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13349 * gst/rtsp-server/rtsp-client.c:
13350 client: don't crash when tunnelid is missing
13351 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13352 don't crash but return an error response to the client.
13355 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13357 * bindings/vala/gst-rtsp-server-0.10.vapi:
13358 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13359 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13360 bindings: update vala bindings with new method
13362 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13364 * gst/rtsp-server/rtsp-session-pool.c:
13365 * gst/rtsp-server/rtsp-session-pool.h:
13366 sessionpool: add function to filter sessions
13367 Add generic function to retrieve/remove sessions.
13369 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13372 configure: bump core/base requirements to release
13374 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13376 * gst/rtsp-server/rtsp-media.c:
13377 media: fix indentation
13379 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13381 * gst/rtsp-server/rtsp-media.c:
13382 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13384 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13386 * gst/rtsp-server/rtsp-media.c:
13387 set state and remove elements of media in for loop
13389 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13391 * bindings/vala/gst-rtsp-server-0.10.vapi:
13392 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13393 Added gst_rtsp_media_remove_elements function to Vala bindings
13395 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13397 * gst/rtsp-server/rtsp-media.c:
13398 * gst/rtsp-server/rtsp-media.h:
13399 Added gst_rtsp_media_remove_elements function
13401 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13403 * gst/rtsp-server/rtsp-media.c:
13404 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13406 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13408 * bindings/vala/gst-rtsp-server-0.10.vapi:
13409 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13410 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13411 Updated Vala bindings
13413 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13415 * gst/rtsp-server/rtsp-media.c:
13416 * gst/rtsp-server/rtsp-media.h:
13417 Added vmethod unprepare to GstRTSPMedia
13418 The default implementation sets the state of the pipeline to GST_STATE_NULL
13420 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13422 * gst/rtsp-server/rtsp-media-factory.c:
13423 * gst/rtsp-server/rtsp-media-factory.h:
13424 Made collect_streams function public
13426 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13428 * gst/rtsp-server/rtsp-media-factory.c:
13429 * gst/rtsp-server/rtsp-media-factory.h:
13430 * gst/rtsp-server/rtsp-media.c:
13431 Added vmethod create_pipeline to GstRTSPMediaFactory
13432 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13434 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13436 * gst/rtsp-server/rtsp-client.c:
13437 client: use g_source_destroy()
13438 We need to use g_source_destroy() because we might have added the source to a
13439 different main context than the default one.
13441 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13443 * gst/rtsp-server/Makefile.am:
13444 * gst/rtsp-server/rtsp-client.c:
13445 * gst/rtsp-server/rtsp-params.c:
13446 * gst/rtsp-server/rtsp-params.h:
13447 rtsp: prepare for handling GET/SET_PARAMETER
13448 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13450 Fix return codes of handlers.
13452 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13454 * gst/rtsp-server/rtsp-media.c:
13455 media: don't leak session pads
13457 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13459 * gst/rtsp-server/rtsp-media.c:
13460 media: clean up the messages a bit
13462 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13464 * gst/rtsp-server/rtsp-sdp.c:
13465 sdp: warn and skip streams without media
13467 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13469 * bindings/vala/gst-rtsp-server-0.10.vapi:
13470 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13471 vala: Fixed typo in header file of RTSPMediaStream
13473 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13475 * gst/rtsp-server/rtsp-media.c:
13477 Fix a debug message
13478 Make dumping RTCP stats configurable
13480 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13482 * gst/rtsp-server/rtsp-media.c:
13483 media: be less verbose and leak less
13485 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13487 * gst/rtsp-server/rtsp-media.c:
13488 media: don't leak the destination address
13490 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13492 * gst/rtsp-server/rtsp-client.c:
13493 * gst/rtsp-server/rtsp-media.c:
13494 * gst/rtsp-server/rtsp-media.h:
13495 * gst/rtsp-server/rtsp-session.c:
13496 * gst/rtsp-server/rtsp-session.h:
13497 rtsp: use RTCP to keep the session alive
13498 Use the RTCP rtcp-from stats field to find the associated session and use this
13499 to keep the session alive.
13501 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13503 * gst/rtsp-server/rtsp-session.c:
13504 session: add 5sec to the real session timeout
13505 Allow the session to live 5sec longer before really timing out. This should give
13506 clients some extra time to keep the session active.
13508 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13510 * gst/rtsp-server/rtsp-client.c:
13511 client: replay OK to GET/SET_PARAMETER
13512 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13513 so that we return OK for those requests.
13515 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13517 * gst/rtsp-server/rtsp-media.c:
13518 * gst/rtsp-server/rtsp-media.h:
13519 media: keep track of active transports
13520 Keep track of which transport is active to avoid closing the connection too
13522 Remove the destination transport also when going to NULL.
13523 Print some stats about the SDES and other RTCP messages we receive from the
13526 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13528 * examples/.gitignore:
13529 * examples/Makefile.am:
13530 * examples/test-sdp.c:
13531 example: add SDP relay example
13533 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13535 * gst/rtsp-server/rtsp-media.c:
13536 media: also count active TCP connections
13538 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13540 * gst/rtsp-server/rtsp-media-factory.c:
13541 * gst/rtsp-server/rtsp-media.c:
13542 * gst/rtsp-server/rtsp-media.h:
13543 rtsp: add support for dynamic elements
13544 Add support for dynamic elements.
13545 Don't set live pipelines back to paused.
13547 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13549 * gst/rtsp-server/rtsp-sdp.c:
13550 sdp: don't add encoding name when absent in caps
13552 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13554 * gst/rtsp-server/rtsp-client.c:
13555 client: warn when we can't do RTP-Info
13557 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13559 * gst/rtsp-server/rtsp-media-factory.c:
13560 factory: factor out the stream construction
13562 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13564 * gst/rtsp-server/rtsp-client.c:
13565 client: only add RTP-Info when we have the info
13566 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13569 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13572 back to development
13574 === release 0.10.3 ===
13576 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13580 - Fixes a bug where it put the wrong verion in pkgconfig
13581 - Link RTP and RTCP sources
13583 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13585 * gst/rtsp-server/rtsp-media.c:
13586 * gst/rtsp-server/rtsp-media.h:
13587 media: link the RTP udpsrc to the session manager
13588 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13589 shut down when the client sends a packet to open firewalls.
13591 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13593 * pkgconfig/gst-rtsp-server.pc.in:
13594 Don't use hard-coded version number in pkg-config file
13596 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13599 back to development
13601 === release 0.10.2 ===
13603 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13608 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13611 * common/m4/.gitignore:
13612 * examples/.gitignore:
13613 * pkgconfig/.gitignore:
13614 add some .gitignore files
13616 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13618 * gst/rtsp-server/rtsp-media.c:
13619 media: seek to key frames
13621 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13623 * gst/rtsp-server/rtsp-media.c:
13624 media: emit the unprepared signal by id
13625 Emit the unprepared signal by id instead of name and set the media as
13628 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13630 * gst/rtsp-server/rtsp-media.c:
13631 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13633 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13635 * gst/rtsp-server/rtsp-server.c:
13636 Added finalize function to GstRTPSPServer to unref session pool and media mapping
13638 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13640 * bindings/vala/gst-rtsp-server-0.10.vapi:
13641 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13642 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13643 Updated vala bindings
13645 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13647 * gst/rtsp-server/Makefile.am:
13648 * gst/rtsp-server/rtsp-client.c:
13649 * gst/rtsp-server/rtsp-media.c:
13650 server: use appsink and appsrc with the API
13651 Use the appsink/appsrc API instead of the signals for higher
13654 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13656 * examples/test-ogg.c:
13657 tests: set the payload type correctly
13659 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13661 * gst/rtsp-server/rtsp-media-factory.c:
13662 factory: connect to the unprepare signal
13663 Connect to the unprepare signal for non-reusable media so that we can remove
13664 them from the cache.
13666 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13668 * gst/rtsp-server/rtsp-media.c:
13669 * gst/rtsp-server/rtsp-media.h:
13670 media: add signal to notify of unprepare
13672 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13674 * gst/rtsp-server/rtsp-media.c:
13675 * gst/rtsp-server/rtsp-media.h:
13676 media: more work on making the media shared
13677 Add a reusable flag to medias, indicating that they can be reused after a state
13681 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13683 * examples/test-readme.c:
13684 examples: mark the example as shared for testing
13686 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13688 * gst/rtsp-server/rtsp-media.c:
13689 * gst/rtsp-server/rtsp-media.h:
13690 client: support shared media
13691 Always perform the state actions even if the target state of the pipeline is
13692 already correct, we still want to add/remove the transports when we are dealing
13694 Keep a counter of the number of active transports for a media so that we can use
13695 this to perform a state change when needed.
13696 Perform a state change of the pipeline only when the first transport was added
13697 or when there are no active transports.
13699 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13701 * gst/rtsp-server/rtsp-client.c:
13702 client: fix refcounting crasher
13703 Don't need to remove the weak refs in the finalize methods, they are already
13704 removed in the dispose.
13705 Don't register the callback with a DestroyNofity.
13707 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13709 * gst/rtsp-server/rtsp-client.c:
13710 Fix rtsp client refcount management in TCP mode.
13711 Don't unref a client ref we never had. Fixes an unref
13712 of an already-free client object after a client
13713 teardown request for me.
13715 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13717 * gst/rtsp-server/rtsp-session.c:
13718 docs: fix typo in API docs
13720 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13722 * gst/rtsp-server/rtsp-media.c:
13723 More seeking fixes.
13724 Keep the udp sources in playing even if we go to paused. unlock the sources when
13726 Add some more debug info.
13727 Only seek when we need to.
13728 Keep track of the position when we go to paused.
13730 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13732 * gst/rtsp-server/rtsp-client.c:
13733 * gst/rtsp-server/rtsp-media.c:
13734 * gst/rtsp-server/rtsp-media.h:
13735 Add beginnings of seeking.
13736 Parse the Range header and perform a seek on the pipeline for the requested
13737 position. It's disabled currently until I figure out what's going wrong.
13739 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13741 * gst/rtsp-server/rtsp-client.c:
13742 allow pause requests for now.
13745 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13747 * gst/rtsp-server/rtsp-client.c:
13748 Remove weak ref on the session in teardown
13749 We need to remove our weakref from the session when we do a teardown because
13750 else we close the TCP connection prematurely.
13752 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13754 * gst/rtsp-server/rtsp-client.c:
13755 * gst/rtsp-server/rtsp-client.h:
13756 * gst/rtsp-server/rtsp-session-pool.c:
13757 Do some more session cleanup
13758 Make session timeout kill the TCP connection that currently watches the
13760 Remove the client timeout property.
13762 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13764 * gst/rtsp-server/rtsp-client.c:
13765 * gst/rtsp-server/rtsp-client.h:
13766 * gst/rtsp-server/rtsp-media.c:
13767 * gst/rtsp-server/rtsp-media.h:
13768 * gst/rtsp-server/rtsp-server.c:
13769 * gst/rtsp-server/rtsp-session.c:
13770 * gst/rtsp-server/rtsp-session.h:
13772 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
13775 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13777 * examples/Makefile.am:
13778 * examples/test-launch.c:
13779 Add example server that takes launch lines
13780 Add an example server that streams any -launch line.
13782 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13784 * examples/test-readme.c:
13785 * gst/rtsp-server/rtsp-client.c:
13786 * gst/rtsp-server/rtsp-media.c:
13787 * gst/rtsp-server/rtsp-media.h:
13788 Add support for live streams
13789 Add support for live streams and ranges
13790 Start on handling TCP data transfer.
13792 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13794 * gst/rtsp-server/rtsp-media.c:
13795 Free the pipeline before other things
13798 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13800 * gst/rtsp-server/rtsp-client.c:
13801 Only free the pending tunnel if there is one
13804 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13806 * gst/rtsp-server/rtsp-client.c:
13807 * gst/rtsp-server/rtsp-client.h:
13808 * gst/rtsp-server/rtsp-media.c:
13809 rtsp-server: Add support for tunneling
13810 Add support for tunneling over HTTP.
13811 Use new connection methods to retrieve the url.
13812 Dispatch messages based on the message type instead of blindly
13813 assuming it's always a request.
13814 Keep track of the watch id so that we can remove it later.
13815 Set the media pipeline to NULL before unreffing the pipeline.
13817 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13819 * gst/rtsp-server/rtsp-client.c:
13820 * gst/rtsp-server/rtsp-client.h:
13821 Fix for channel -> watch rename in gstreamer
13822 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
13824 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13826 * gst/rtsp-server/rtsp-client.c:
13827 * gst/rtsp-server/rtsp-client.h:
13829 Use the async RTSP channels instead of spawning a new thread for each client.
13830 If a sessionid is specified in a request, fail if we don't have the session.
13832 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13834 * gst/rtsp-server/rtsp-media.c:
13835 Add better debug info
13836 Add some better debug info.
13838 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13840 * examples/test-video.c:
13842 Add support for session timeouts in the example.
13844 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13846 * gst/rtsp-server/rtsp-session-pool.c:
13847 * gst/rtsp-server/rtsp-session-pool.h:
13848 Pass GTimeVal around for performance reasons
13849 Get the current time only once and pass it around so that sessions don't have to
13850 get the current time anymore.
13851 Add experimental support for a GSource that dispatches when the session needs to
13854 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13856 * gst/rtsp-server/rtsp-session.c:
13857 * gst/rtsp-server/rtsp-session.h:
13858 Add better support for session timeouts
13859 Add a method to request the number of milliseconds when a session will timeout.
13861 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13863 * gst/rtsp-server/rtsp-media.c:
13864 * gst/rtsp-server/rtsp-media.h:
13865 Add suport for RTP manager monitoring
13866 Add the first stage in monitoring the rtp manager.
13867 Make sure we don't update the state to something we don't want.
13869 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13871 * gst/rtsp-server/rtsp-client.c:
13872 Add support for session keepalive
13873 Get and update the session timeout for all requests. get the session as early as
13876 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13878 * gst/rtsp-server/rtsp-media-factory.h:
13879 * gst/rtsp-server/rtsp-media.c:
13880 * gst/rtsp-server/rtsp-media.h:
13881 Handle media bus messages
13882 Handle media bus messages in a custom mainloop and dispatch them to the
13883 RTSPMedia objects. Let the default implementation handle some common messages.
13885 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13887 * gst/rtsp-server/rtsp-client.c:
13888 * gst/rtsp-server/rtsp-session-pool.c:
13889 * gst/rtsp-server/rtsp-session.c:
13890 Some more session timeout handling
13891 Move the session header setting code to a central place so that we always add
13892 the timeout parameter too.
13893 Handle timeouts by running the session cleanup code.
13894 Stop media before cleaning up.
13896 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13898 * gst/rtsp-server/rtsp-client.c:
13899 * gst/rtsp-server/rtsp-client.h:
13900 Add timeout property
13901 Add a timeout property ot the client and make the other properties into GObject
13904 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13906 * gst/rtsp-server/rtsp-session-pool.c:
13907 Use getters and setters in property code
13908 Use the getters and setters for the timeout property instead of locking
13911 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13913 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
13915 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13917 * gst/rtsp-server/rtsp-session-pool.c:
13918 * gst/rtsp-server/rtsp-session-pool.h:
13919 * gst/rtsp-server/rtsp-session.c:
13920 * gst/rtsp-server/rtsp-session.h:
13921 Add more timeout stuff
13922 Add method to check if a session is expired.
13923 Add method to perform cleanup on a session pool.
13925 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13927 * gst/rtsp-server/rtsp-client.c:
13928 * gst/rtsp-server/rtsp-session-pool.c:
13929 * gst/rtsp-server/rtsp-session-pool.h:
13930 * gst/rtsp-server/rtsp-session.c:
13931 * gst/rtsp-server/rtsp-session.h:
13932 Add beginnings of session timeouts and limits
13933 Add the timeout value to the Session header for unusual timeout values.
13934 Allow us to configure a limit to the amount of active sessions in a pool. Set a
13935 limit on the amount of retry we do after a sessionid collision.
13936 Add properties to the sessionid and the timeout of a session. Keep track of
13937 creation time and last access time for sessions.
13939 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13941 * gst/rtsp-server/rtsp-client.c:
13942 * gst/rtsp-server/rtsp-media.c:
13943 * gst/rtsp-server/rtsp-media.h:
13944 * gst/rtsp-server/rtsp-sdp.c:
13945 * gst/rtsp-server/rtsp-session-pool.c:
13946 * gst/rtsp-server/rtsp-session.c:
13947 * gst/rtsp-server/rtsp-session.h:
13948 Cleanup of sessions and more
13949 Fix the refcounting of media and sessions in the client. Properly clean up the
13950 session data when the client performs a teardown.
13951 Add Server header to responses.
13952 Allow for multiple uri setups in one session.
13953 Add Range header to the PLAY response and add the range attribute to the SDP
13955 Fix the session pool remove method, it used the wrong key in the hashtable. Also
13956 give the ownership of the sessionid to the session object.
13958 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13960 * gst/rtsp-server/rtsp-server.c:
13961 * gst/rtsp-server/rtsp-server.h:
13963 Rename the 'server_port' variable to simply 'port'.
13965 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13968 * gst/rtsp-server/rtsp-client.c:
13969 * gst/rtsp-server/rtsp-media.c:
13970 * gst/rtsp-server/rtsp-media.h:
13971 * gst/rtsp-server/rtsp-session.c:
13972 * gst/rtsp-server/rtsp-session.h:
13973 Rework the way we handle transports for streams
13974 Make the media accept an array of transports for the streams that we have
13975 configured for the play/pause requests.
13976 Implement server states for a client and its media.
13977 Require 0.10.22.1 (git HEAD) of gstreamer.
13979 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13981 * gst/rtsp-server/rtsp-client.c:
13982 * gst/rtsp-server/rtsp-media-factory.c:
13983 Drop const from functions dealing with urls
13984 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
13985 have the right const in them.
13987 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13989 * gst/rtsp-server/rtsp-client.c:
13990 * gst/rtsp-server/rtsp-media.c:
13991 * gst/rtsp-server/rtsp-sdp.c:
13995 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13997 * gst/rtsp-server/rtsp-client.c:
13998 * gst/rtsp-server/rtsp-media-factory.c:
13999 * gst/rtsp-server/rtsp-media.c:
14000 * gst/rtsp-server/rtsp-media.h:
14002 Don't keep a reference to the GstRTSPMedia in the stream.
14003 Free more things when freeing the GstRTSPMedia.
14005 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14008 * gst/rtsp-server/rtsp-media-factory.c:
14009 * gst/rtsp-server/rtsp-media-factory.h:
14010 * gst/rtsp-server/rtsp-media.c:
14011 * gst/rtsp-server/rtsp-media.h:
14012 * gst/rtsp-server/rtsp-server.c:
14013 * gst/rtsp-server/rtsp-server.h:
14014 More docs and small cleanups
14015 Add some more docs and update the README
14016 Cleanup some method names.
14017 Remove an unneeded idx field in the GstRTSPMediaStream
14019 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14022 * examples/Makefile.am:
14023 * examples/test-readme.c:
14024 Add a README and more example code
14025 Add a README file that contains a small introduction on how to use the server
14026 along with the example code explained in the readme.
14028 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14030 * gst/rtsp-server/rtsp-media.c:
14031 * gst/rtsp-server/rtsp-server.c:
14032 Fix some leaks and change default port
14033 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14034 we finished the initial preroll. If we keep them locked, setting the pipeline to
14035 NULL will not stop and clean up the sources correctly.
14036 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14038 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14040 * gst/rtsp-server/rtsp-session.c:
14041 * gst/rtsp-server/rtsp-session.h:
14042 Cleanups to the session object
14043 Remove some unneeded variables in the session state of a stream such as the
14044 owner media and the server transport.
14045 Get the configuration of a media stream in a session based on the media_stream
14046 in the original object instead of our cached index.
14047 Free more data in the finalize method.
14049 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14051 * gst/rtsp-server/rtsp-client.c:
14052 * gst/rtsp-server/rtsp-client.h:
14053 Cleanups and reuse media from DESCRIBE
14054 Handle thread create errors.
14055 Rename some internal methods to better match what they actually do.
14056 Handle misconfiguration of session_pool and media_mapping gracefully.
14057 Cache the DESCRIBE media and uri in the client connection and reuse them when
14058 we receive a SETUP request in the same connection for the same uri.
14059 Cleanup the client connection object.
14061 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14063 * gst/rtsp-server/rtsp-media-factory.c:
14064 * gst/rtsp-server/rtsp-media-factory.h:
14065 * gst/rtsp-server/rtsp-media.c:
14066 * gst/rtsp-server/rtsp-media.h:
14067 Add shared properties to media and factory
14068 Add the shared property to media.
14069 Implement some simple caching in the factory depending on if the media is shared
14072 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14074 * gst/rtsp-server/rtsp-client.c:
14075 Add a little comment
14076 Add some comment about the content-base header.
14078 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14080 * examples/Makefile.am:
14081 * examples/test-mp4.c:
14082 * examples/test-ogg.c:
14083 * examples/test-video.c:
14084 * gst/rtsp-server/Makefile.am:
14085 * gst/rtsp-server/rtsp-client.c:
14086 * gst/rtsp-server/rtsp-client.h:
14087 * gst/rtsp-server/rtsp-media-factory.c:
14088 * gst/rtsp-server/rtsp-media-factory.h:
14089 * gst/rtsp-server/rtsp-media.c:
14090 * gst/rtsp-server/rtsp-media.h:
14091 * gst/rtsp-server/rtsp-sdp.c:
14092 * gst/rtsp-server/rtsp-sdp.h:
14093 * gst/rtsp-server/rtsp-server.c:
14094 * gst/rtsp-server/rtsp-server.h:
14095 * gst/rtsp-server/rtsp-session.c:
14096 * gst/rtsp-server/rtsp-session.h:
14097 Reorganize things, prepare for media sharing
14098 Added various other test server examples
14099 Move the SDP message generation to a separate helper.
14100 Refactor common code for finding the session.
14101 Add content-base for realplayer compatibility
14102 Clean up request uris before processing for better vlc compatibility.
14103 Move prerolling and pipeline construction to the RTSPMedia object.
14104 Use multiudpsink for future pipeline reuse.
14106 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14109 Back to development
14112 === release 0.10.1 ===
14114 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14117 Make 0.10.1 release
14120 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14122 * bindings/vala/Makefile.am:
14124 Add more directories and files to the dist.
14126 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14128 * bindings/python/Makefile.am:
14129 * bindings/python/rtspserver.override:
14130 Fixed compile error of python bindings
14132 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14134 * bindings/vala/gst-rtsp-server-0.10.vapi:
14135 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14136 Marked values as nullable accordingly
14138 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14140 * bindings/vala/gst-rtsp-server-0.10.vapi:
14141 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14142 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14143 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14144 Updated Vala bindings
14146 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14148 * gst/rtsp-server/rtsp-client.c:
14149 * gst/rtsp-server/rtsp-media-mapping.c:
14150 * gst/rtsp-server/rtsp-media-mapping.h:
14151 * gst/rtsp-server/rtsp-media.h:
14152 * gst/rtsp-server/rtsp-session-pool.h:
14153 Cleanups and doc updates
14154 Add some more documentation and do some minor cleanups here and there.
14156 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14158 * gst/rtsp-server/rtsp-client.c:
14159 * gst/rtsp-server/rtsp-media-factory.c:
14160 * gst/rtsp-server/rtsp-media-factory.h:
14161 * gst/rtsp-server/rtsp-media.c:
14162 * gst/rtsp-server/rtsp-media.h:
14163 * gst/rtsp-server/rtsp-session.c:
14164 * gst/rtsp-server/rtsp-session.h:
14166 Rename GstRTSPMediaBin to GstRTSPMedia
14167 Parse the request url into a GstRTSPUri object and pass this object to the
14168 various handlers and methods that require the uri.
14170 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14174 Add some more docs and remove some old code from the example.
14176 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14178 * gst/rtsp-server/rtsp-client.c:
14179 Handle state change failures better
14180 Handle state change failures better when changing the state of the pipeline to
14183 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14185 * gst/rtsp-server/rtsp-media-factory.c:
14186 * gst/rtsp-server/rtsp-media-factory.h:
14187 Make element creation more extendible
14188 Add get_element vmethod to the default MediaFactory so that subclasses can just
14189 override that method and still use the default logic for making a MediaBin from
14192 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14195 * gst/rtsp-server/Makefile.am:
14196 * gst/rtsp-server/rtsp-client.c:
14197 * gst/rtsp-server/rtsp-client.h:
14198 * gst/rtsp-server/rtsp-media-factory.c:
14199 * gst/rtsp-server/rtsp-media-factory.h:
14200 * gst/rtsp-server/rtsp-media-mapping.c:
14201 * gst/rtsp-server/rtsp-media-mapping.h:
14202 * gst/rtsp-server/rtsp-media.c:
14203 * gst/rtsp-server/rtsp-media.h:
14204 * gst/rtsp-server/rtsp-server.c:
14205 * gst/rtsp-server/rtsp-server.h:
14206 * gst/rtsp-server/rtsp-session.c:
14207 * gst/rtsp-server/rtsp-session.h:
14208 Make the server handle arbitrary pipelines
14209 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14210 The GstMediaBin object has a handle to a bin with elements and to a list of
14211 GstMediaStream objects that this bin produces.
14212 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14213 with methods to register and remove those mappings.
14214 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14215 used by the server instance.
14216 Modify the example application so that it shows how to create custom pipelines
14217 attached to a specific mount point.
14218 Various misc cleanps.
14220 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14222 * gst/rtsp-server/rtsp-server.c:
14223 * gst/rtsp-server/rtsp-server.h:
14224 Allow setting a custom media factory for a server
14226 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14228 * gst/rtsp-server/rtsp-client.c:
14229 * gst/rtsp-server/rtsp-client.h:
14230 Allow setting a custom media factory for a client.
14232 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14234 * gst/rtsp-server/Makefile.am:
14235 Add Makefile entry for the media factory
14237 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14239 * gst/rtsp-server/rtsp-media-factory.c:
14240 * gst/rtsp-server/rtsp-media-factory.h:
14241 Add media factory to map urls to media pipeline objects.
14243 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14245 * gst/rtsp-server/rtsp-media.c:
14246 * gst/rtsp-server/rtsp-media.h:
14247 Add comments. Remove unused field
14249 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14251 * gst/rtsp-server/rtsp-session-pool.c:
14252 * gst/rtsp-server/rtsp-session-pool.h:
14253 Allow custom session pools to override the session id allocation algorithms Add some comments.
14255 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14257 * gst/rtsp-server/rtsp-session.h:
14260 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14262 * gst/rtsp-server/rtsp-client.c:
14263 * gst/rtsp-server/rtsp-client.h:
14264 Move the connection code in one place Add some comments
14266 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14268 * gst/rtsp-server/rtsp-server.c:
14269 * gst/rtsp-server/rtsp-server.h:
14270 Make vmethod to create and accept new clients. Add some docs.
14272 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14274 * gst/rtsp-server/rtsp-server.c:
14275 * gst/rtsp-server/rtsp-server.h:
14276 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14278 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14280 * gst/rtsp-server/rtsp-client.c:
14281 * gst/rtsp-server/rtsp-client.h:
14282 Name the parameters more appropriately.
14284 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14286 * gst/rtsp-server/rtsp-session-pool.c:
14287 Do some more cleanup of the session pool.
14289 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14291 * gst/rtsp-server/Makefile.am:
14292 * gst/rtsp-server/rtsp-client.c:
14293 Check if return value of gst_rtsp_session_get_media is not NULL
14295 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14297 * gst/rtsp-server/Makefile.am:
14298 Install rtsp-session and rtsp-session-pool headers
14300 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14305 * bindings/python/Makefile.am:
14306 * bindings/python/arg-types.py:
14307 * bindings/python/codegen/Makefile.am:
14308 * bindings/python/codegen/__init__.py:
14309 * bindings/python/codegen/argtypes.py:
14310 * bindings/python/codegen/code-coverage.py:
14311 * bindings/python/codegen/codegen.py:
14312 * bindings/python/codegen/definitions.py:
14313 * bindings/python/codegen/defsparser.py:
14314 * bindings/python/codegen/docextract.py:
14315 * bindings/python/codegen/docgen.py:
14316 * bindings/python/codegen/fileprefix.override:
14317 * bindings/python/codegen/fileprefixmodule.c:
14318 * bindings/python/codegen/h2def.py:
14319 * bindings/python/codegen/mergedefs.py:
14320 * bindings/python/codegen/mkskel.py:
14321 * bindings/python/codegen/override.py:
14322 * bindings/python/codegen/reversewrapper.py:
14323 * bindings/python/codegen/scmexpr.py:
14324 * bindings/python/rtspserver-types.defs:
14325 * bindings/python/rtspserver.defs:
14326 * bindings/python/rtspserver.override:
14327 * bindings/python/rtspservermodule.c:
14329 Add python bindings.
14331 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14333 * bindings/Makefile.am:
14335 Don't go into python dir when requirements for python bindings are missing
14337 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14339 * bindings/Makefile.am:
14340 * bindings/vala/Makefile.am:
14342 Install Vala bindings if vala is available
14344 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14346 * bindings/vala/gst-rtsp-server-0.10.deps:
14347 * bindings/vala/gst-rtsp-server-0.10.vapi:
14348 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14349 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14350 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14351 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14352 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14353 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14354 Regenerated Vala bindings
14356 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14358 * bindings/vala/gst-rtsp-server.vapi:
14359 * bindings/vala/packages/gst-rtsp-server.metadata:
14360 Fixed typo in included headers for vala bindings
14362 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14366 * pkgconfig/Makefile.am:
14367 * pkgconfig/gst-rtsp-server.pc.in:
14368 Added pkgconfig file
14370 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14372 * bindings/vala/gst-rtsp-server.vapi:
14373 * bindings/vala/packages/gst-rtsp-server.excludes:
14374 * bindings/vala/packages/gst-rtsp-server.gi:
14375 * bindings/vala/packages/gst-rtsp-server.metadata:
14376 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14378 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14380 * bindings/vala/gst-rtsp-server.vapi:
14381 * bindings/vala/packages/gst-rtsp-server.deps:
14382 * bindings/vala/packages/gst-rtsp-server.files:
14383 * bindings/vala/packages/gst-rtsp-server.gi:
14384 * bindings/vala/packages/gst-rtsp-server.metadata:
14385 * bindings/vala/packages/gst-rtsp-server.namespace:
14386 Added Vala bindings
14388 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14390 * gst/rtsp-server/rtsp-session.c:
14391 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14393 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14395 * examples/Makefile.am:
14396 * gst/rtsp-server/Makefile.am:
14397 Put GStreamer version in library name
14399 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14401 * examples/Makefile.am:
14402 * gst/rtsp-server/Makefile.am:
14403 Fix some issues to pass distcheck
14405 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14407 * gst/rtsp-server/rtsp-server.c:
14408 Added port property to GstRTSPServer class.
14410 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14415 * examples/Makefile.am:
14418 * gst/rtsp-server/Makefile.am:
14419 * gst/rtsp-server/rtsp-client.c:
14420 * gst/rtsp-server/rtsp-client.h:
14421 * gst/rtsp-server/rtsp-media.c:
14422 * gst/rtsp-server/rtsp-media.h:
14423 * gst/rtsp-server/rtsp-server.c:
14424 * gst/rtsp-server/rtsp-server.h:
14425 * gst/rtsp-server/rtsp-session-pool.c:
14426 * gst/rtsp-server/rtsp-session-pool.h:
14427 * gst/rtsp-server/rtsp-session.c:
14428 * gst/rtsp-server/rtsp-session.h:
14430 Split in library and example program
14432 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14434 * src/rtsp-client.h:
14435 Removed obsolete variable
14437 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14439 * src/rtsp-client.c:
14440 * src/rtsp-client.h:
14441 Removed pipeline variable GstRTSPClient, because it's only used in one function
14443 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14445 * src/rtsp-media.c:
14446 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14448 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14450 * src/rtsp-session.c:
14451 Initialize some more vars.
14453 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14455 * src/rtsp-session.c:
14456 Initialize variable to avoid compiler warning.
14458 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14461 Add a reasonable generic .gitignore