1 === release 1.19.90 ===
3 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
7 * docs/gst_plugins_cache.json:
8 * gst-rtsp-server.doap:
12 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
15 Update ChangeLogs for 1.19.90
17 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
19 * examples/test-appsrc2.c:
20 gst-rtsp-server: Fix leak in appsrc2 example
21 In the need-data appsrc callback, a buffer is pulled from the
22 appsink. This buffer is then copied so that metadata is writable.
23 The copy is pushed to the appsrc but it doesn't take ownership
24 of the buffer so we need to manually unref it. The original buffer
25 is finally unreffed when the sample is freed.
26 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
28 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
32 meson: Add explicit check: kwarg to all run_command() calls
33 This is required since Meson 0.61.0, and causes a warning to be
35 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
36 https://github.com/mesonbuild/meson/issues/9300
37 This exposed a bunch of places where we had broken run_command()
38 calls, unnecessary run_command() calls, and places where check: true
40 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
42 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
44 * gst/rtsp-server/meson.build:
45 rtsp-server: add gst_dep to gst_rtsp_server_deps
46 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
47 will avoid the following build failure, because the correct girdir
48 location will be retrieved from gstreamer-1.0.pc:
49 /home/giuliobenetti/autobuild/run/instance-3/output-1/host/riscv32-buildroot-linux-gnu/sysroot/usr/bin/g-ir-compiler gst/rtsp-server/GstRtspServer-1.0.gir --output gst/rtsp-server/GstRtspServer-1.0.typelib --includedir=/usr/share/gir-1.0
50 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
51 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
52 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
53 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
55 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
56 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
58 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
60 * gst/rtsp-server/rtsp-stream.c:
61 rtsp-stream: fix get_rates raciness
62 Prior to this patch, we considered that a stream was blocking
63 whenever a pad probe was triggered for either the RTP pad or
65 This led to situations where we subsequently unblocked and expected
66 to find a segment on the RTP pad, which was racy.
67 Instead, we now only consider that the stream is blocking when
68 the pad probe for the RTP pad has triggered with a blockable object
69 (buffer, buffer list, gap event).
70 The RTCP pad is simply blocked without affecting the state of the
73 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
75 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
77 * docs/gst_plugins_cache.json:
81 === release 1.19.3 ===
83 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
88 * docs/gst_plugins_cache.json:
89 * gst-rtsp-server.doap:
93 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
96 Update ChangeLogs for 1.19.3
98 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
101 meson: require matching GStreamer dep versions for unstable development releases
102 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
103 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
105 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
107 * tests/check/meson.build:
108 meson: update for meson.build_root() and .build_source() deprecation
109 -> use meson.project_build_root() or .global_build_root() instead.
110 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
112 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
115 * tests/check/meson.build:
116 meson: update for dep.get_pkgconfig_variable() deprecation
117 ... in favour of dep.get_variable('foo', ..) which in some
118 cases allows for further cleanups in future since we can
119 extract variables from pkg-config dependencies as well as
120 internal dependencies using this mechanism.
121 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
123 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
125 * gst/rtsp-server/meson.build:
126 * gst/rtsp-sink/meson.build:
127 rtsp-server: define G_LOG_DOMAIN
129 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
131 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
134 meson: bump meson requirement to >= 0.59
135 For monorepo build and ugly/bad, for advanced feature
136 option API like get_option('xyz').required(..) which
137 we use in combination with the 'gpl' option.
138 For rest of modules for consistency (people will likely
139 use newer features based on the top-level requirement).
140 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
142 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
145 meson: Streamline the way we detect when to build documentation
146 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
148 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
151 * gst/rtsp-server/meson.build:
153 meson: List libraries and their corresponding gir definition
154 Introduces a `libraries` variable that contains all libraries in a
155 list with the following format:
159 'lib': library_object
160 'gir': [ {full gir definition in a dict } ]
165 It therefore refactors the way we build the gir so that we can reuse the
166 same information to build them against 'gstreamer-full' in gst-build
167 when linking statically
168 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
170 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
172 * gst/rtsp-server/meson.build:
173 meson: Mark files as files()
174 Making it more robust and future proof
175 And fix issues that it creates
176 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
178 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
180 * gst/rtsp-server/rtsp-media.c:
181 rtsp-media: Unprepare suspended medias too
182 Previously suspended medias immediately reached the UNPREPARED state
183 without going through the media's unprepare() vfunc. This didn't allow
184 the media subclass to do any additional cleanup, and for example the
185 shutdown-eos property of GstRTSPMedia was ignored.
186 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
188 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
190 * gst/rtsp-server/rtsp-media.c:
191 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
192 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
194 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
196 * gst/rtsp-server/rtsp-client.c:
197 * gst/rtsp-server/rtsp-session.c:
198 * gst/rtsp-server/rtsp-session.h:
199 rtsp-client: make sure sessmedia will not get freed while used
200 handle_*_request() functions were all retrieving the session media from
201 the session by calling gst_rtsp_session_get_media () which is a transfer-none
202 call. If a session timeout happens at that time, the session media may get freed
203 making the pointer invalid..
205 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
207 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
209 * gst/rtsp-server/rtsp-media.c:
210 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
211 Previously the status was only changed for other medias.
212 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
214 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
216 * gst/rtsp-server/rtsp-session.c:
217 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
218 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
219 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
221 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
224 doc: update IRC links to OFTC
225 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
227 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
229 * docs/gst_plugins_cache.json:
232 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
234 === release 1.19.2 ===
236 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
241 * docs/gst_plugins_cache.json:
242 * gst-rtsp-server.doap:
246 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
248 * gst/rtsp-server/rtsp-media.c:
249 * gst/rtsp-server/rtsp-stream.c:
250 * gst/rtsp-server/rtsp-stream.h:
251 * gst/rtsp-sink/gstrtspclientsink.c:
252 Protection against early RTCP packets.
253 When receiving RTCP packets early the funnel is not ready yet and
254 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
255 This causes the thread that handle RTCP packets to go to pause mode.
256 Since this thread is in pause mode there will be no further callbacks to
257 handle keep-alive for incoming RTCP packets. This will make the session
258 time out if the client is not using another keep-alive mechanism.
259 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
260 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
262 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
266 Update COPYING.LIB, COPYING files
267 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
269 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
271 * docs/gst_plugins_cache.json:
275 === release 1.19.1 ===
277 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
282 * docs/gst_plugins_cache.json:
283 * gst-rtsp-server.doap:
287 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
289 * gst/rtsp-server/rtsp-stream.c:
290 rtsp-stream: use new gst_buffer_new_memdup()
291 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
293 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
295 * gst/rtsp-server/rtsp-media-factory-uri.c:
296 rtsp-media: fix leak when adding converter
297 Free the previous caps before reusing the variable for the converter caps.
298 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
300 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
302 * gst/rtsp-server/rtsp-client.c:
303 rtsp-client: fix leak adding headers
304 gst_rtsp_message_add_header() makes a copy of the header, instead
306 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
308 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
310 * gst/rtsp-server/rtsp-stream.c:
311 Use gst_element_request_pad_simple...
312 Instead of the deprecated gst_element_get_request_pad.
313 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
315 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
317 * gst/rtsp-server/rtsp-media.c:
318 rtsp-media: Ensure the bus watch is removed during unprepare
319 It's possible for the destruction of the source to be delayed.
320 Instead of relying on the dispose() to remove the bus watch, do
322 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
324 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
327 docs: minor spelling correction in README
328 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
330 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
332 * examples/test-replay-server.c:
333 test-replay-server: minor spelling corrections
334 Bumped on these while investigating the example code.
335 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
337 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
339 * tests/check/gst/stream.c:
340 tests: Don't fail tests if IPv6 not available.
341 On computers with IPv6 disabled it shouldn't result in a test failure.
342 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
344 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
346 * gst/rtsp-server/rtsp-media.c:
347 rtsp-media: Add one more case to seek avoidance
348 This is an extension to the previous commit. There can also be cases where the
349 start position is not specified, in those cases we should also avoid doing
350 seeking unless it's forced.
351 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
353 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
355 * gst/rtsp-server/rtsp-media.c:
356 rtsp-media: Improve skipping trickmode seek.
357 We can also skip the seek if the end range is already
359 Avoids initial seek on play start if playing full stream.
360 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
362 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
364 * gst/rtsp-sink/gstrtspclientsink.c:
365 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
366 It's sufficient to run them during the FIRST stage instead of in both.
367 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
369 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
371 * tests/check/gst/rtspclientsink.c:
372 tests: rtspclientsink: fix some leaks
373 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
375 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
377 * gst/rtsp-sink/gstrtspclientsink.c:
378 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
379 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
381 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
383 * tests/check/gst/rtspclientsink.c:
384 rtspclientsink: add unit test for potential shutdown deadlock
385 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
387 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
389 * gst/rtsp-sink/gstrtspclientsink.c:
390 rtspclientsink: fix deadlock on shutdown before preroll
391 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
392 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
394 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
396 * gst/rtsp-server/rtsp-stream.c:
397 rtsp-stream: avoid deadlock in send_func
398 Currently the send_func() runs in a thread of its own which is started
399 the first time we enter handle_new_sample(). It runs in an outer loop
400 until priv->continue_sending is FALSE, which happens when a TEARDOWN
401 request is received. We use a local variable, cont, which is initialized
402 to TRUE, meaning that we will always enter the outer loop, and at the
403 end of the outer loop we assign it the value of priv->continue_sending.
404 Within the outer loop there is an inner loop, where we wait to be
405 signaled when there is more data to send. The inner loop is exited when
406 priv->send_cookie has changed value, which it does when more data is
407 available or when a TEARDOWN has been received.
408 But if we get a TEARDOWN before send_func() is entered we will get stuck
409 in the inner loop because no one will increase priv->session_cookie
411 By not entering the outer loop in send_func() if priv->continue_sending
412 is FALSE we make sure that we do not get stuck in send_func()'s inner
413 loop should we receive a TEARDOWN before the send thread has started.
414 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
415 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
417 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
419 * gst/rtsp-server/rtsp-client.c:
420 rtsp-client: cleanup transports during TEARDOWN
421 When tunneling RTP over RTSP the stream transports are stored in a hash
422 table in the GstRTSPClientPrivate struct. They are used for, among other
423 things, mapping channel id to stream transports when receiving data from
424 the client. The stream tranports are created and added to the hash table
425 in handle_setup_request(), but unfortuately they are not removed in
426 handle_teardown_request(). This means that if the client sends data on
427 the RTSP connection after it has sent the TEARDOWN, which is often the
428 case when audio backchannel is enabled, handle_data() will still be able
429 to map the channel to a session transport and pass the data along to it.
430 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
431 because the stream is no longer joined to a bin.
432 We avoid this by removing the stream transports from the hash table when
433 we handle the TEARDOWN request.
434 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
436 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
438 * docs/gst_plugins_cache.json:
439 * gst/rtsp-sink/gstrtspclientsink.c:
440 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
441 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
443 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
445 * tests/check/gst/client.c:
446 Add test cases for mountpoint of '/'
447 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
449 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
451 * gst/rtsp-server/rtsp-client.c:
452 * gst/rtsp-server/rtsp-mount-points.c:
453 * gst/rtsp-server/rtsp-session-media.c:
454 Make a mount point of "/" work correctly.
455 As far as I can tell, this is neither explicitly allowed nor
456 forbidden by RFC 7826.
457 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
458 use in the wild (presumably with non-GStreamer servers).
459 GStreamer's prior behavior was confusing, in that
460 gst_rtsp_mount_points_add_factory() would appear to accept a mount
461 path of "" or "/", but later connection attempts would fail with a
462 "media not found" error.
463 This commit makes a mount path of "/" work for either form of URL,
464 while an empty mount path ("") is rejected and logs a warning.
465 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
467 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
469 * docs/gst_plugins_cache.json:
470 * gst/rtsp-sink/gstrtspclientsink.c:
471 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
472 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
474 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
476 * gst/rtsp-server/rtsp-media.c:
477 rtsp-media: Only count senders when counting blocked streams
478 Only sender streams sends the GstRTSPStreamBlocking message, so only
479 these should be counted before setting media status to prepared.
480 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
482 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
484 * gst/rtsp-sink/gstrtspclientsink.c:
485 rtspclientsink add proper support for uri queries
486 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
488 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
490 * gst/rtsp-server/rtsp-client.c:
491 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
492 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
493 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
495 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
497 * gst/rtsp-server/rtsp-stream.c:
498 rtsp-stream: collect a clock_rate when blocking
499 This lets us provide a clock_rate in a fashion similar to the
500 other code paths in get_rtpinfo()
501 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
503 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
505 * gst/rtsp-server/rtsp-media.c:
506 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
507 Otherwise this will cause memory corruption as the property expects a 64
509 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
511 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
513 * gst/rtsp-server/rtsp-media.c:
514 * gst/rtsp-server/rtsp-stream.c:
515 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
516 To prevent cases with prerolling when the inactive stream prerolls first
517 and the server proceeds without waiting for the active stream, we will
518 ignore GstRTSPStreamBlocking messages from incomplete streams. When
519 there are no complete streams (during DESCRIBE), we will listen to all
521 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
523 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
525 * tests/check/gst/media.c:
526 * tests/check/meson.build:
527 * tests/files/test.avi:
528 media test: Add test for seeking one active stream with a demuxer
529 Add another seek_one_active_stream test but with a demuxer. The demuxer
530 will flush both streams in opposed to the existing test which only
531 flushes the active stream. This will help exposing problems with the
532 prerolling process after a flushing seek.
533 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
535 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
537 * gst/rtsp-server/meson.build:
539 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
540 * pkgconfig/gstreamer-rtsp-server.pc.in:
541 * pkgconfig/meson.build:
542 Meson: Use pkg-config generator
543 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
545 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
548 meson: update glib minimum version to 2.56
549 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
551 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
553 * examples/test-launch.c:
554 * gst/rtsp-server/rtsp-media-factory.c:
555 * gst/rtsp-server/rtsp-media-factory.h:
556 * gst/rtsp-server/rtsp-media.c:
557 * gst/rtsp-server/rtsp-server-internal.h:
558 * gst/rtsp-server/rtsp-stream.c:
559 * tests/check/gst/client.c:
560 rtsp-media-factory: expose API to disable RTCP
561 This is supported by the RFC, and can be useful on systems where
562 allocating two consecutive ports is problematic, and RTCP is not
564 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
566 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
568 * hooks/pre-commit.hook:
570 git: use our standard pre commit hook
571 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
573 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
575 * gst/rtsp-server/rtsp-stream.c:
576 rtsp-stream: make use of blocked_running_time in query_position
577 When blocking, the sink element will not have received a buffer
578 yet and the position query will fail. Instead, we make use of
579 the running time of the buffer we blocked on.
580 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
582 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
584 * gst/rtsp-server/rtsp-stream.c:
585 rtsp-stream: collect rtp info when blocking
586 We don't unblock the stream anymore before replying to the
587 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
588 so the sinks don't have a last-sample after potentially flush
589 seeking. seek_trickmode waits for preroll however, which means
590 the stream will block and wait for a first buffer. Subsequent
591 calls to get_rtpinfo() can thus make use of the information.
592 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
593 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
595 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
597 * examples/meson.build:
598 * examples/test-replay-server.c:
599 * examples/test-replay-server.h:
600 examples: Add an example for loop playback
601 This demo example shows a way of file loop playback of a given source.
602 Note that client seek request is not properly implemented yet.
603 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
605 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
607 * gst/rtsp-server/rtsp-media.c:
608 rtsp-media: Plug memory leak
609 The get-storage signal of rtpbin increases the ref count of the storage.
610 So we have to unref it after usage.
611 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
613 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
615 * gst/rtsp-server/rtsp-media.c:
616 rtsp-media: Get rates only on sender streams
617 When play a media with both sender and receiver stream, like ONVIF
618 back channel audio in, gst_rtsp_media_get_rates call
619 gst_rtsp_stream_get_rates for each stream to set the rates. But
620 gst_rtsp_stream_get_rates return false for the receiver steam, which
621 lead a g_assert crash.
622 Instead to get rates on all streams, now just get rates on sender
624 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
626 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
628 * gst/rtsp-server/rtsp-media.c:
629 * gst/rtsp-server/rtsp-server-internal.h:
630 * gst/rtsp-server/rtsp-stream.c:
631 rtsp-media: set a 0 storage size for TCP receivers
632 ulpfec correction is obviously useless when receiving a stream
633 over TCP, and in TCP modes the rtp storage receives non
634 timestamped buffers, causing it to queue buffers indefinitely,
635 until the queue grows so large that sanity checks kick in and
636 warnings start to get emitted.
637 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
639 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
641 * gst/rtsp-server/rtsp-stream.c:
642 rtsp-stream: preroll on gap events
643 This allows negotiating a SDP with all streams present, but only
644 start sending packets at some later point in time
645 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
647 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
649 * gst/rtsp-server/rtsp-media.c:
650 rtsp-media: do not unblock on unsuspend
651 rtsp_media_unsuspend() is called from handle_play_request()
652 before sending the play response. Unblocking the streams here
653 was causing data to be sent out before the client was ready
654 to handle it, with obvious side effects such as initial packets
655 getting discarded, causing decoding errors.
656 Instead we can simply let the media streams be unblocked when
657 the state of the media is set to PLAYING, which occurs after
658 sending the play response.
659 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
661 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
664 ci: include template from gst-ci master branch again
666 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
668 * docs/gst_plugins_cache.json:
672 === release 1.18.0 ===
674 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
680 * docs/gst_plugins_cache.json:
681 * gst-rtsp-server.doap:
685 === release 1.17.90 ===
687 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
692 * docs/gst_plugins_cache.json:
693 * gst-rtsp-server.doap:
697 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
699 * gst/rtsp-server/rtsp-thread-pool.c:
700 rtsp-thread-pool.c: fix clang 10 warning
701 clang 10 is complaining about incompatible types due to the
704 ../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]
706 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
708 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
710 * gst/rtsp-server/rtsp-thread-pool.c:
711 rtsp-thread-pool.c: fix clang 10 warning
712 clang 10 is complaining about incompatible types due to the
715 ../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]
717 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
719 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
721 * gst/rtsp-server/rtsp-sdp.c:
722 rtsp-sdp: Fix resource leak in mikey messsage
723 Fixed a resource leak for mikey message while adding crypto session
725 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
727 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
730 * scripts/extract-release-date-from-doap-file.py:
731 meson: set release date from .doap file for releases
732 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
734 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
736 * gst/rtsp-server/rtsp-stream.c:
737 rtsp-stream: explicitly set caps on udpsrc elements
738 This causes them to send caps events before data flow, which is
739 usually a pretty correct thing to do!
740 Not doing so manifested in a bug where ssrcdemux wouldn't forward
741 the caps it had received with an extra ssrc field, as it hadn't
742 received any caps event.
744 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
746 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
748 * docs/gst_plugins_cache.json:
752 === release 1.17.2 ===
754 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
759 * docs/gst_plugins_cache.json:
760 * gst-rtsp-server.doap:
764 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
766 * docs/gst_plugins_cache.json:
767 doc: Stop documenting properties from parents
769 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
771 * docs/gst_plugins_cache.json:
772 docs: Fix version in the plugins cache
773 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
775 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
777 * gst/rtsp-sink/gstrtspclientsink.c:
778 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
779 It's deprecated, unneeded and doesn't do anything anymore.
780 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
782 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
787 === release 1.17.1 ===
789 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
794 * docs/gst_plugins_cache.json:
795 * gst-rtsp-server.doap:
799 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
801 * gst/rtsp-server/rtsp-media.c:
802 rtsp-media: Add/configure transports when completing the pipeline
803 Otherwise the transports are not set up yet during the PLAY request
804 handling when unsuspending (and thus unblocking) the media.
805 In case of live pipelines this then causes the first few packets to go
806 to the sinks before they know what to do with them, and they simply
807 discard them which is rather suboptimal in case of keyframes.
808 For non-live pipelines this is not a problem because the sink will still
809 be PAUSED and as such not send out the data yet but wait until it goes
810 to PLAYING, which is late enough.
811 Adding the transports multiple times is not a problem: if the transport
812 is already added it won't be added another time and TRUE will be
814 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
816 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
817 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
819 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
821 * gst/rtsp-server/rtsp-media.c:
822 rtsp-media: Fix misleading comment
823 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
825 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
827 * gst/rtsp-server/rtsp-media.c:
828 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
829 The pad probes are not needed anymore at this point and later when
830 reaching buffering 100% only the state is changed, no unblocking
832 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
834 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
836 * gst/rtsp-server/rtsp-media.c:
837 rtsp-media: Remove duplicated media_unblock() function
838 It does literally the same as media_streams_set_blocked(FALSE).
839 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
841 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
843 * examples/test-onvif-server.c:
844 test-onvif-server: cast ntp-offset property value to 64 bit
845 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
847 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
849 * docs/gst_plugins_cache.json:
850 docs: Update plugins cache
852 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
854 * examples/test-onvif-server.c:
855 * examples/test-onvif-server.h:
856 * gst/rtsp-server/rtsp-onvif-media-factory.h:
857 onvif-media-factory: define autoptr cleanup function
858 And have the factory in the onvif-server example inherit from
859 GstRTSPOnvifMediaFactory.
860 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
862 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
864 * docs/gst_plugins_cache.json:
865 docs: Update plugins cache
867 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
869 * tests/check/gst/rtspserver.c:
870 tests: enforce I420 format
871 Test was not enforcing a video format on videotestsrc. I420 was picked as it
872 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
873 true (gst-plugins-base!689).
874 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
876 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
878 * gst/rtsp-sink/gstrtspclientsink.c:
879 plugins: uddate gst_type_mark_as_plugin_api() calls
881 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
884 doc: Require hotdoc >= 0.11.0
886 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
888 * docs/gst_plugins_cache.json:
889 docs: Update gst_plugins_cache.json
891 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
893 * gst/rtsp-sink/gstrtspclientsink.c:
894 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
896 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
898 * gst/rtsp-server/meson.build:
899 meson: gir: remove bogus sources_top_dir kwarg
900 Doesn't actually exist. Was fixed differently in Meson
901 so that the user doesn't have to specify it.
902 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
904 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
906 * tests/check/meson.build:
907 tests: put registry into tests/check not the gst/ subdir
908 Underscorify the test name before setting GST_REGISTRY,
909 so the registry actually ends up in the current build dir
911 For consistency with the other modules, but should also
912 avoid problems on windows.
913 Also fix indentation of environment block.
914 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
916 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
918 * tests/check/meson.build:
919 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
920 If core is built as a subproject (e.g. as in gst-build), make sure to use
921 the gst-plugin-scanner from the built subproject. Without this, gstreamer
922 might accidentally use the gst-plugin-scanner from the install prefix if
923 that exists, which in turn might drag in gst library versions we didn't
924 mean to drag in. Those gst library versions might then be older than
925 what our current build needs, and might cause our newly-built plugins
926 to get blacklisted in the test registry because they rely on a symbol
927 that the wrongly-pulled in gst lib doesn't have.
928 This should fix running of unit tests in gst-build when invoking
929 meson test or ninja test from outside the devenv for the case where
930 there is an older or different-version gst-plugin-scanner installed
931 in the install prefix.
932 In case no gst-plugin-scanner is installed in the install prefix, this
933 will fix "GStreamer-WARNING: External plugin loader failed. This most
934 likely means that the plugin loader helper binary was not found or
935 could not be run. You might need to set the GST_PLUGIN_SCANNER
936 environment variable if your setup is unusual." warnings when running
938 In the case where we find GStreamer core via pkg-config we use
939 a newly-added pkg-config var "pluginscannerdir" to get the right
940 directory. This has the benefit of working transparently for both
941 installed and uninstalled pkg-config files/setups.
942 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
944 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
946 * tests/check/meson.build:
947 tests: gst-plugins-base and -bad plugins are required for the unit tests
948 Make hard requirement until we have more fine-grained control
949 in the unit tests. Of course the presence of the .pc file doesn't
950 imply that the plugins we need are actually there, but it's at
951 least a step in the right direction.
952 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
954 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
956 * tests/check/meson.build:
957 tests: pick up rtsp-server plugins from build directory only
958 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
960 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
962 * gst/rtsp-server/rtsp-media.c:
963 rtsp-media: wait for all GstRTSPStreamBlocking messages
964 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
965 each active stream when checking if all streams are blocked.
966 Without this change there will be a race condition when using two or
967 more streams and rtsp-media receives a GstRTSPStreamBlocking message
968 from one of the streams. This is because rtsp-media then checks if all
969 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
970 stream. This function call returns TRUE if the stream has sent a
971 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
972 receive this message. This would then result in that rtsp-media
973 erroneously thinks it is blocking all streams which could result in
974 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
975 preroll, this could result in that rtsp-media thinks that the pipeline
976 is prerolled even though that might not be the case.
977 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
979 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
981 * gst/rtsp-server/rtsp-media.c:
982 rtsp-media: update expected_async_done during suspend
983 Set expected_async_done to FALSE in default_suspend() if a state change
984 occurs and the return value from set_target_state() is something other
985 than GST_STATE_CHANGE_ASYNC.
986 Without this change there is a risk that expected_async_done will be
987 TRUE even though no asynchronous state change is taking place. This
988 could happen if the pipeline is set to PAUSED using
989 media_set_pipeline_state_locked(), an asynchronous state change starts
990 and then the media is suspended (which could result in a state change,
991 aborting the asynchronous state change). If the media is suspended
992 before the asynchronous state change ends then expected_async_done will
993 be TRUE but no asynchronous state change is taking place.
994 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
996 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
998 * gst/rtsp-server/rtsp-client.c:
999 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1000 There was a race condition where client was being finalized and
1001 concurrently in some other thread the rtsp ctrl timout was relying on
1002 client data that was being freed.
1003 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1004 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1006 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1008 * gst/rtsp-server/rtsp-media-factory.c:
1009 * gst/rtsp-server/rtsp-media-factory.h:
1010 * gst/rtsp-server/rtsp-media.c:
1011 * gst/rtsp-server/rtsp-media.h:
1012 media-factory: complete DSCP QoS setting support
1013 add dscp_qos setting support at factory and media level to setup IP DSCP
1014 field of bounded UDP sinks.
1015 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1016 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1018 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1020 * gst/rtsp-server/rtsp-client.c:
1021 rtsp-client: Fix some race conditions around timeout source removal
1022 We always need to take the lock while accessing it as otherwise another
1023 thread might've removed it in the meantime. Also when destroying and
1024 creating a new one, ensure that the mutex is not shortly unlocked in
1025 between as during that time another one might potentially be created
1027 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1029 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1031 * gst/rtsp-server/rtsp-media.c:
1032 * gst/rtsp-server/rtsp-stream.c:
1033 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1034 And the same for gst_rtsp_stream_get_rates().
1035 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1037 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1039 * examples/test-onvif-server.c:
1040 examples: test-onvif-server: fix compiler warnings on raspbian
1041 Fix printf format for 64-bit variables.
1042 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1044 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1046 * gst/rtsp-server/rtsp-stream-transport.c:
1047 * gst/rtsp-server/rtsp-stream-transport.h:
1048 * gst/rtsp-server/rtsp-stream.c:
1049 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1050 The old API is preserved now and new API was added that provides the
1051 additional parameter to the callback.
1052 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1053 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1055 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1057 * gst/rtsp-server/rtsp-client.c:
1058 rtsp-client: Store the timeout source by pointer instead of id
1059 That way we don't have to retrieve it again from the main context when
1060 destroying it but can directly do so.
1061 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1063 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1065 * gst/rtsp-server/rtsp-client.c:
1066 rtsp-client: Clean up watch/watch context and related state consistently
1067 And assert that it was cleaned up properly before the client is
1068 finalized. If something is still around when the client is shut down
1069 then something went very wrong before.
1070 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1072 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1074 * gst/rtsp-server/rtsp-client.c:
1075 * tests/check/gst/rtspserver.c:
1076 rtsp-client: Combine the pre-session and post-session timeout
1077 They previously used the same state but different mechanisms and
1078 functions, which was difficult to follow, error prone and simply
1080 Also adjust the test for the post-session timeout a bit to be less racy
1081 now that the timing has slightly changed.
1082 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1084 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1086 * gst/rtsp-server/rtsp-client.c:
1087 rtsp-client: Don't ever close the client connection directly when a session is torn down
1088 There might be other sessions that are running over the same RTSP
1089 connection and we should not simply close the client directly if one of
1091 By default the connection will be closed once the client closes it or
1092 the OS does. This behaviour can be adjusted with the
1093 post-session-timeout property, which allows to close it automatically
1094 from the server side after all sessions are gone and the given timeout
1096 This reverts the previous commit.
1097 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1099 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1101 * gst/rtsp-server/rtsp-client.c:
1102 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1103 Instead of closing it never at all. Previously there was only code that
1104 closed the client asynchronously if sending the response happened
1105 asynchrously at a later time.
1106 Thanks to Christian M for debugging this issue.
1107 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1108 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1110 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1112 * gst/rtsp-server/rtsp-stream.c:
1113 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1114 Otherwise no sink is found for multicast sreams and the less accurate
1115 fallback is used to determine the current sequence number and timestamp.
1117 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1119 * gst/rtsp-server/rtsp-auth.c:
1120 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1121 When using the basic authentication scheme, we wouldn't validate that
1122 the authorization field of the credentials is not NULL and pass it on
1123 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1124 dereference the NULL pointer and crash.
1125 A specially crafted (read: invalid) RTSP header can cause this to
1127 As a solution, check for the authorization to be not NULL before
1128 continuing processing it and if it is simply fail authentication.
1129 This fixes CVE-2020-6095 and TALOS-2020-1018.
1130 Discovered by Peter Wang of Cisco ASIG.
1132 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1134 * gst/rtsp-server/rtsp-client.c:
1135 rtsp-client: Use watch_context before unref
1136 Move the usage of priv->watch_context to beginning of function
1137 gst_rtsp_client_finalize. Instead of use it after
1138 g_main_context_unref (priv->watch_context).
1140 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1142 * gst/rtsp-server/rtsp-stream.c:
1143 rtsp-stream: fix deadlock on transport removal
1144 We cannot take the RTSPStream lock while holding a transport backlog
1145 lock, as remove_transport may be called externally, which will
1146 take first the RTSPStream lock then the transport backlog lock.
1148 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1150 * gst/rtsp-server/rtsp-server-internal.h:
1151 * gst/rtsp-server/rtsp-stream-transport.c:
1152 * gst/rtsp-server/rtsp-stream.c:
1153 rtsp-stream: clear backlog when removing transport
1154 This ensures we don't end up calling any of transports' callbacks
1155 with a potentially unreffed user_data (in practice, a client that
1156 may have been removed)
1158 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1160 * gst/rtsp-server/rtsp-stream.c:
1161 rtsp-stream: marshal calls to send_tcp_message to a single thread
1162 In order to address the race condition pointed out at
1163 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1164 we get rid of the send thread pool, and instead spawn and manage
1165 a single thread to pull samples from app sinks and add them to
1166 the transport's backlogs.
1167 Additionally, we now also always go through the backlogs in order
1168 to simplify the logic.
1170 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1172 * gst/rtsp-server/rtsp-server-internal.h:
1173 * gst/rtsp-server/rtsp-stream-transport.c:
1174 * gst/rtsp-server/rtsp-stream.c:
1175 rtsp-stream: properly protect TCP backlog access
1177 We cannot hold stream->lock while pushing data, but need
1178 to consistently check the state of the backlog both from
1179 the send_tcp_message function and the on_message_sent function,
1180 which may or may not be called from the same thread.
1181 This commit introduces internal API to allow for potentially
1182 recursive locking of transport streams, addressing a race
1183 condition where the RTSP stream could push items out of order
1184 when popping them from the backlog.
1186 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1188 * gst/rtsp-server/rtsp-media.c:
1189 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1190 It's taken ownership of by the media, and returned with `transfer none`
1191 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1192 first then any bindings will wrongly take ownership of the pipeline once
1193 it arrives in bindings code.
1195 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1197 * examples/test-onvif-client.c:
1198 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1200 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1202 * gst/rtsp-server/rtsp-media.c:
1203 rtsp-media: fix default latency
1205 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1207 * gst/rtsp-server/rtsp-client.c:
1208 rtsp-client: make closing more thread safe
1209 + Take the watch lock prior to using priv->watch
1210 + Flush both the watch and connection before closing / unreffing
1211 gst_rtsp_connection_close() is not threadsafe on its own, this is
1212 a workaround at the client level, where we control both the watch
1215 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1217 * gst/rtsp-server/rtsp-latency-bin.c:
1218 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1221 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1222 `your_type_get_instance_private()` function instead
1225 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1227 * gst/rtsp-server/rtsp-client.c:
1228 * tests/check/gst/rtspserver.c:
1229 rtsp-client: add property post-session-timeout
1230 This is a TCP connection timeout for client connections, in seconds.
1231 If a positive value is set for this property, the client connection
1232 will be kept alive for this amount of seconds after the last session
1233 timeout. For negative values of this property the connection timeout
1234 handling is delegated to the system (just as it was before).
1237 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1239 * gst/rtsp-server/rtsp-stream.c:
1240 rtsp-stream: check for NULL transports prior to ref'ing
1242 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1244 * gst/rtsp-server/rtsp-server-internal.h:
1245 * gst/rtsp-server/rtsp-stream-transport.c:
1246 * gst/rtsp-server/rtsp-stream.c:
1247 rtsp-stream: fix checking of TCP backpressure
1248 The internal index of our appsinks, while it can be used to
1249 determine whether a message is RTP or RTCP, is not necessarily
1250 the same as the interleaved channel. Let the stream-transport
1251 determine the channel to check backpressure for, the same way
1252 it determines the channel according to whether it is sending
1255 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1257 * gst/rtsp-server/rtsp-session.c:
1258 rtsp-session: Butcher the file to please gst-indent in the CI
1259 This should be reverted once the CI has an updated gst-indent.
1261 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1263 * gst/rtsp-server/rtsp-session.c:
1264 * gst/rtsp-server/rtsp-session.h:
1265 * gst/rtsp-sink/gstrtspclientsink.c:
1266 * gst/rtsp-sink/gstrtspclientsink.h:
1267 rtsp-session & client: Remove deprecated GTimeVal
1268 GTimeVal won't work past 2038
1270 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1272 * gst/rtsp-server/rtsp-auth.c:
1273 rtsp-auth: fix default token leak
1275 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1277 * gst/rtsp-sink/gstrtspclientsink.c:
1278 gstrtspclientsink: unref transports when closing bin
1281 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1283 * gst/rtsp-server/rtsp-media.c:
1284 rtsp-media: Force seek when flush flag is set
1285 The commit "rtsp-client: define all seek accuracy flags from
1286 setup_play_mode" changed the behaviour of when doing a seek.
1287 Before that commit, having the flush flag set would result in a seek
1289 Even if no seek was needed. One reason to force seek is to flush old buffers
1290 created in Describe requests.
1291 Thus adding force seek also for flush flag will result in play request
1294 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1296 * gst/rtsp-server/rtsp-client.c:
1297 rtsp-client: Revitalize dead code
1298 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1301 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1303 * gst/rtsp-server/rtsp-sdp.c:
1304 rtsp-sdp: Don't try to use non-initialized values
1305 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1306 returns TRUE. Also avoid the whole clock signalling block if we're not
1307 dealing with senders.
1312 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1314 * gst/rtsp-server/rtsp-stream-transport.c:
1315 * gst/rtsp-server/rtsp-stream.c:
1316 * tests/check/gst/stream.c:
1317 rtsp-stream: Removing invalid transports returns false
1318 When removing transports an assertion was that the transports passed in
1319 for removal are present in the list, however that can't be assumed.
1320 As an example if a transport was removed from a thread running
1321 send_tcp_message, the main thread can try to remove the same transport
1322 again if it gets a handle_pause_request. This will not effect the
1323 transport list but it will effect n_tcp_transports as it will be
1324 decrement and then have the wrong value.
1326 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1328 * tests/check/gst/client.c:
1329 client test: add scale and speed negative tests
1330 Negative tests for scale and speed should be done as well, verify that
1331 the response code is "400 Bad request" when a bad request is done.
1333 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1335 * gst/rtsp-server/rtsp-auth.c:
1336 * gst/rtsp-server/rtsp-client.c:
1337 * gst/rtsp-server/rtsp-media-factory.c:
1338 * gst/rtsp-server/rtsp-media.c:
1339 * gst/rtsp-server/rtsp-server.c:
1340 * gst/rtsp-server/rtsp-session-pool.c:
1341 * gst/rtsp-server/rtsp-stream.c:
1342 * gst/rtsp-sink/gstrtspclientsink.c:
1343 Don't pass default GLib marshallers for signals
1344 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1345 actually internally optimize the signal (if the marshaller is available
1346 in GLib itself) by also setting the valist marshaller. This makes the
1347 signal emission a bit more performant than the regular marshalling,
1348 which still needs to box into `GValue` and call libffi in case of a
1350 Note that for custom marshallers, one would use
1351 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1353 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1355 * gst/rtsp-server/rtsp-mount-points.c:
1356 GstRTSPMountPoints: Remove any existing factory before adding a new one
1357 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1358 previous mount point will be freed" which was true when it was
1359 implemented using a GHashTable. But in 2012 it got rewrote using a
1360 GSequence and since then it could have 2 factories for the same path.
1361 Which one gets used is random, depending on the sorting order of 2
1364 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1366 * gst/rtsp-server/rtsp-client.c:
1367 * gst/rtsp-server/rtsp-server-internal.h:
1368 * gst/rtsp-server/rtsp-stream-transport.c:
1369 * gst/rtsp-server/rtsp-stream-transport.h:
1370 * gst/rtsp-server/rtsp-stream.c:
1371 stream: refactor TCP backpressure handling
1372 The previous implementation stopped sending TCP messages to
1373 all clients when a single one stopped consuming them, which
1374 obviously created problems for shared media.
1375 Instead, we now manage a backlog in stream-transport, and slow
1376 clients are removed once this backlog exceeds a maximum duration,
1377 currently hardcoded.
1380 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1383 meson: build gir even when cross-compiling if introspection was enabled explicitly
1384 This can be made to work in certain circumstances when
1385 cross-compiling, so default to not building g-i stuff
1386 when cross-compiling, but allow it if introspection was
1387 enabled explicitly via -Dintrospection=enabled.
1388 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1390 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1392 * gst/rtsp-server/rtsp-session.c:
1393 rtsp-session: clean up comment extra-timeout
1395 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1397 * gst/rtsp-server/rtsp-client.c:
1398 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1399 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1400 from the RTSP context.
1403 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1405 * gst/rtsp-server/rtsp-client.c:
1406 * gst/rtsp-server/rtsp-media.c:
1407 * gst/rtsp-server/rtsp-media.h:
1408 rtsp-client: Lock shared media
1409 For shared media we got race conditions. Concurrently rtsp clients might
1410 suspend or unsuspend the shared media and thus change the state without
1411 the clients expecting that.
1412 By introducing a lock that can be taken by callers such as rtsp_client
1413 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1414 to handle the media sequentially thus allowing one client to finish its
1415 rtsp call before another client calls on the same media.
1416 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1419 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1421 * gst/rtsp-server/rtsp-session.c:
1422 rtsp-session: add property extra-timeout
1423 Extra time to add to the timeout, in seconds. This only
1424 affects the time until a session is considered timed out
1425 and is not signalled in the RTSP request responses.
1426 Only the value of the timeout property is signalled in the
1429 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1431 * gst/rtsp-server/rtsp-stream.c:
1432 rtsp-stream : fix race condition in send_tcp_message
1433 If one thread is inside the send_tcp_message function and are done
1434 sending rtp or rtcp messages so the n_outstanding variable is zero
1435 however have not exit the loop sending the messages. While sending its
1436 messages, transports have been added or removed to the transport list,
1437 so the cache should be updated. If now an additional thread comes to
1438 the function send_tcp_message and trying to send rtp messages it will
1439 first destroy the rtp cache that is still being iterated trough by the
1443 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1452 * examples/.gitignore:
1453 * examples/Makefile.am:
1455 * gst/rtsp-server/.gitignore:
1456 * gst/rtsp-server/Makefile.am:
1457 * gst/rtsp-sink/Makefile.am:
1458 * pkgconfig/.gitignore:
1459 * pkgconfig/Makefile.am:
1461 * tests/Makefile.am:
1462 * tests/check/Makefile.am:
1463 Remove autotools build
1465 Maybe we can now use the meson pkgconfig module
1466 for .pc files? (Does it support uninstalled now?)
1468 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1470 * tests/check/gst/client.c:
1471 client: fix test mem leak in attach_rate_tweaking_probe
1473 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1475 * tests/check/gst/media.c:
1476 media: remove memleak in test test_media_seek
1478 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1480 * tests/check/gst/rtspserver.c:
1481 rtspserver: Remove memleak in test test_double_play
1483 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1485 * gst/rtsp-server/rtsp-media.c:
1486 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1488 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1490 * gst/rtsp-server/rtsp-media.c:
1491 * tests/check/gst/rtspserver.c:
1492 rtsp-media: Unblock all streams
1493 When unsuspending and going to PLAYING, unblock all streams instead of
1494 only those that are linked (the linked streams are the ones for which
1495 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1496 pushing buffers on unlinked streams.
1497 This change is because playback using single-threaded demuxers like
1498 matroska-demux could be blocked if SETUP was not called for all media.
1499 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1500 gstflvdemux, qtdemux, and matroska-demux) will handle
1501 GST_FLOW_NOT_LINKED automatically.
1504 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1506 * gst/rtsp-server/rtsp-media.c:
1507 * tests/check/gst/rtspserver.c:
1508 rtsp-media: Wait on async when needed.
1509 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1510 In the unit test the pause from adjust_play_mode will cause a preroll
1511 and after that async-done will be produced.
1512 Without this patch there are no one consuming this async-done and when
1513 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1514 wait for async-done. But then it wrongly find the async-done prodused by
1515 adjus_play_mode and continue executing without waiting for the preroll
1518 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1520 * gst/rtsp-server/rtsp-client.c:
1521 rtsp-client: RTP Info when completed_sender
1522 Change condition that should be fulfilled regarding RTPInfo.
1523 Replace !gst_rtsp_media_is_receive_only with
1524 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1525 for a sender pipeline that is complete. Only then a RTPInfo should
1527 gst_rtsp_media_is_receive_only gives different answears depending on
1529 If Describe is called wth URL+options for backchannel SDP will give only
1530 audio and only backchannel a=sendonly
1531 If Describe is called on URL+options that gives both audio and video
1532 direction from server to client, pipelines are created. Thus
1533 receive_only will return false, even though Setup only would setup
1535 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1536 streams are complete.
1538 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1540 * gst/rtsp-server/rtsp-client.c:
1541 * tests/check/gst/client.c:
1542 rtsp-client: RTP Info exists conditionally in PLAY
1543 If RTP Info is missing and it is not a receiver only, eg. audio
1544 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1545 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1546 Since 1.14 there is audio backchannel support. Thus RTP-info is
1547 conditional now. When audio backchannel only mode, there is no RTP-info.
1550 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1552 * examples/test-onvif-client.c:
1553 test-onvif-client: remove unused query
1555 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1557 * gst/rtsp-server/rtsp-client.c:
1558 rtsp-client: RTP Info must exist in PLAY response
1559 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1562 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1564 * examples/test-onvif-client.c:
1565 test-onvif-client: perform accurate seeks
1566 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1567 Also, modify how we compute the position: position queries in
1568 PAUSED mode fail to account for the newly-prerolled frame, leading
1569 to frame skips when performing seeks in that state. Instead,
1570 compute the current position from the last sample.
1572 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1574 * gst/rtsp-server/rtsp-client.c:
1575 * gst/rtsp-server/rtsp-media.c:
1576 * gst/rtsp-server/rtsp-media.h:
1577 * tests/check/gst/rtspserver.c:
1578 Use complete streams for scale and speed.
1579 Without this patch it's always stream0 that is used to get segment event
1580 that is used to set scale and speed. This even if client not doing SETUP
1581 for stream0. At least in suspend mode reset this not working since then
1582 it's just random if send_rtp_sink have got any segment event. There are
1583 no check if send_rtp_sink for stream0 got any data before media is
1584 prerolled after PLAY request.
1586 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1588 * examples/test-onvif-server.c:
1589 * examples/test-onvif-server.h:
1590 examples/onvif-server: fix werror build with clang
1591 ../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]
1592 self->incoming_segment->format, self->incoming_segment->flags,
1593 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1594 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1595 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1597 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1598 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1600 <scratch space>:77:1: note: expanded from here
1603 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1604 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1606 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1607 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1609 <scratch space>:9:1: note: expanded from here
1612 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1613 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1614 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1616 <scratch space>:12:1: note: expanded from here
1620 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1623 meson: Don't generate doc cache when no plugins are enabled
1624 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1626 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1628 * examples/test-onvif-client.c:
1629 test-onvif-client: stdin is not defined in MSVC
1631 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1633 * gst/rtsp-server/rtsp-media.c:
1634 rtsp-media: add missing Since tag
1636 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1638 * examples/test-onvif-client.c:
1639 test-onvif-client: STDIN_FILENO is not portable
1640 If not defined, define it to _fileno(stdin) on Windows, 0
1643 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1645 * examples/test-onvif-server.c:
1646 test-onvif-server: downgrade logging
1648 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1650 * examples/meson.build:
1651 * examples/test-onvif-client.c:
1652 * examples/test-onvif-server.c:
1653 examples: add ONVIF client / server example
1655 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1657 * gst/rtsp-server/rtsp-client.c:
1658 * gst/rtsp-server/rtsp-media.c:
1659 rtsp-client: define all seek accuracy flags from setup_play_mode
1660 We then pass those to adjust_play_mode, which needs to operate
1661 on the "final" seek flags, as previously the code in rtsp-media
1662 was assuming that accuracy seek flags (accurate / key_unit) should
1663 not be set if the flags passed to the seek method were already set.
1665 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1667 * gst/rtsp-server/rtsp-media-factory-uri.c:
1668 * gst/rtsp-server/rtsp-media.c:
1669 rtsp-media: Try to get dynamic payloaders by name from their bin first
1670 First try "pay", then "pay_%s" (where %s == pad name). And only then
1671 fall back to the code that simply takes the first payloader that is
1673 The current code usually works (but is racy) because it will always take
1674 the payloader that was last added (due to g_list_prepend() when adding
1675 elements) in pad-added and that's usually the correct one. But if a new
1676 payloader is added between pad-added and us trying to get it, we would
1677 get the wrong payloader.
1679 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1681 * tests/check/gst/client.c:
1682 client test: expect any port in transport
1683 setup_multicast_client sets a 5000-5010 range for the client
1684 ports, it is incorrect to expect the transport to always use
1688 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1690 * tests/check/gst/onvif.c:
1691 onvif tests: use g_cond_wait() correctly
1692 g_cond_wait() has to be called in a loop until required conditions
1696 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1698 * gst/rtsp-server/rtsp-stream.c:
1699 rtsp-stream: Not wait on receiver streams when pre-rolling
1700 Without this patch there are problem pre-rolling when using audio back
1702 Without this patch a probe will be created for all streams including
1703 the stream for audio backchannel. To pre-roll all this pads have to
1704 receive data. Since the stream for audio backchannel is a receiver this
1706 The solution is to never create any probes for streams that are for
1707 incomming data and instead set them as blocking already from beginning.
1709 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1711 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1712 * gst/rtsp-server/rtsp-onvif-media.c:
1713 onvif-media: fix "void function returning a value" compiler warning
1715 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1717 * gst/rtsp-server/rtsp-media.c:
1718 rtsp-media: make sure streams are blocked when sending seek
1719 The recent ONVIF work exposed a race condition when dealing with
1720 multiple streams: one of the sinks may preroll before other streams
1721 have started flushing. This led to the pipeline posting async-done
1722 prematurely, when some streams were actually still in the middle
1723 of performing a flushing seek. The newly-added code looks up a
1724 sticky segment event on the first stream in order to respond to
1725 the PLAY request with accurate Scale and Speed headers. In the
1726 failure condition, the first stream was flushing, and thus had
1727 no sticky segment event, leading to the PLAY request failing,
1728 and in turn the test.
1730 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
1733 * gst/rtsp-server/rtsp-media-factory-uri.h:
1736 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1738 * gst/rtsp-server/rtsp-client.c:
1739 * gst/rtsp-server/rtsp-client.h:
1740 * gst/rtsp-server/rtsp-media.c:
1741 * gst/rtsp-server/rtsp-media.h:
1742 * gst/rtsp-server/rtsp-onvif-client.c:
1743 * gst/rtsp-server/rtsp-onvif-client.h:
1744 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1745 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1746 * gst/rtsp-server/rtsp-onvif-media.c:
1747 * gst/rtsp-server/rtsp-onvif-server.h:
1748 * gst/rtsp-server/rtsp-stream.c:
1749 * gst/rtsp-server/rtsp-stream.h:
1750 * tests/check/gst/media.c:
1751 * tests/check/gst/onvif.c:
1752 * tests/check/meson.build:
1753 onvif: Implement and test the Streaming Specification
1754 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
1756 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1758 * gst/rtsp-server/rtsp-client.c:
1759 * gst/rtsp-server/rtsp-client.h:
1760 rtsp-client: add gst_rtsp_client_get_stream_transport()
1761 This will be used in the onvif tests in order to validate the
1762 data transmitted over TCP: for streaming to continue after a
1763 data message has been provided to client->send_func, the client
1764 is responsible for marking the message as sent on the relevant
1767 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1769 * gst/rtsp-server/rtsp-client.c:
1770 client: Scale implies TRICK_MODE
1772 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1774 * gst/rtsp-server/rtsp-client.c:
1775 client: compare booleans, not pointers to them
1777 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
1779 * gst/rtsp-server/rtsp-media.c:
1780 * gst/rtsp-server/rtsp-stream.c:
1781 * tests/check/gst/media.c:
1782 Reverse playback support
1783 GStreamer plays segment from stop to start when doing reverse playback.
1784 RTSP implies that media should be played from start of Range header to
1785 its stop. Hence we swap start and stop times before passing them to
1787 Also make gst_rtsp_stream_query_stop always return value that can be
1788 used as stop time of Range header.
1790 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
1792 * gst/rtsp-server/rtsp-client.c:
1793 * gst/rtsp-server/rtsp-media.c:
1794 * gst/rtsp-server/rtsp-media.h:
1795 * tests/check/gst/client.c:
1796 rtsp-client: add support for Scale and Speed header
1797 Add support for the RTSP Scale and Speed headers by setting the rate in
1798 the seek to (scale*speed). We then check the resulting segment for rate
1799 and applied rate, and use them as values for the Speed and Scale headers
1801 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1803 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
1805 * gst/rtsp-server/rtsp-client.c:
1806 * gst/rtsp-server/rtsp-client.h:
1807 rtsp-client: allow sub classes to adjust the seek
1808 Adds a new virtual function, adjust_play_mode(), that allows
1809 sub classes to adjust the seek done on the media. The sub class can
1810 modify the values of the the seek flags and the rate.
1811 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1813 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
1815 * gst/rtsp-server/rtsp-media.c:
1816 * gst/rtsp-server/rtsp-media.h:
1817 * gst/rtsp-server/rtsp-stream.c:
1818 * gst/rtsp-server/rtsp-stream.h:
1819 * tests/check/gst/media.c:
1820 rtsp-media: allow specifying rate when seeking
1821 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
1822 caller to specify the rate for the seek. Also added functions in
1823 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
1824 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1826 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
1830 meson: Bump minimal GLib version to 2.44
1831 This means we can use some newer features and get rid of some
1832 boilerplate code using the G_DECLARE_* macros.
1833 As discussed on IRC, 2.44 is old enough by now to start depending on it.
1835 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1837 * docs/libs/.gitignore:
1838 * docs/libs/Makefile.am:
1839 * docs/libs/gst-rtsp-server-docs.sgml:
1840 * docs/libs/gst-rtsp-server-sections.txt:
1841 * docs/libs/gst-rtsp-server.types:
1842 docs: remove obsolete gtk-doc related files
1844 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1846 * gst/rtsp-sink/gstrtspclientsink.c:
1847 doc: remove xml from comments
1849 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
1851 * docs/gst_plugins_cache.json:
1853 docs: Stop building the doc cache by default
1854 And update the cache
1855 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
1857 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
1859 * docs/gst_plugins_cache.json:
1860 docs: Update plugins documentation cache
1862 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
1865 * gst/rtsp-server/rtsp-context.c:
1866 * gst/rtsp-server/rtsp-session-pool.c:
1867 doc: Fix some docstrings
1869 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
1875 * docs/gst_plugins_cache.json:
1878 * docs/plugin-index.md:
1879 * docs/plugin-sitemap.txt:
1882 * docs/version.entities.in:
1883 * gst/rtsp-server/meson.build:
1884 * gst/rtsp-sink/meson.build:
1886 * meson_options.txt:
1887 docs: Port to hotdoc
1889 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1891 * gst/rtsp-server/rtsp-auth.c:
1892 * gst/rtsp-server/rtsp-client.h:
1893 rtsp-server: Fix various Since markers
1895 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1897 * gst/rtsp-server/rtsp-media.c:
1898 * gst/rtsp-server/rtsp-sdp.c:
1899 * gst/rtsp-server/rtsp-session-media.c:
1900 * gst/rtsp-server/rtsp-stream.c:
1901 rtsp-server: Add various Since: 1.14 markers
1903 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1905 * gst/rtsp-server/rtsp-media-factory.c:
1906 * gst/rtsp-server/rtsp-media.c:
1907 * gst/rtsp-server/rtsp-stream-transport.c:
1908 * gst/rtsp-server/rtsp-stream.c:
1909 rtsp-server: Add various missing Since: 1.16 markers
1911 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
1913 * gst/rtsp-sink/gstrtspclientsink.c:
1914 rtspclientsink: Set async-handling=false for the internal bins
1915 Without this we can easily run into a race condition with async state changes:
1916 - the pipeline is doing an async state change
1917 - we set the internal bins to PLAYING but that's ignored because an
1918 async state change is currently pending
1919 - the async state change finishes but does not change the state of the
1920 internal bins because of locked_state==TRUE
1921 - the internal bins stay in PAUSED forever
1923 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1925 * gst/rtsp-sink/gstrtspclientsink.c:
1926 rtspclientsink: Use write_messages() API to send buffer lists in one go
1927 And to write messages with multiple memories also via writev().
1929 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
1931 * gst/rtsp-server/rtsp-client.c:
1932 * gst/rtsp-server/rtsp-client.h:
1933 * gst/rtsp-server/rtsp-server-object.h:
1934 * gst/rtsp-server/rtsp-server.c:
1935 rtsp-client: Handle Content-Length limitation
1936 Add functionality to limit the Content-Length.
1937 API addition, Enhancement.
1938 Define an appropriate request size limit and reject requests
1939 exceeding the limit with response status 413 Request Entity Too Large
1942 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1949 === release 1.16.0 ===
1951 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1957 * gst-rtsp-server.doap:
1961 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
1963 * gst/rtsp-sink/gstrtspclientsink.c:
1964 rtspclientsink: Notify the stream transport about each written message
1965 Otherwise it will never try to send us the next one: it tries to keep
1966 exactly one message in-flight all the time.
1967 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
1968 in the client sink we always write data out synchronously.
1970 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
1972 * gst/rtsp-server/rtsp-stream.c:
1973 rtsp_server: Free thread pool before clean transport cache
1974 If not waiting for free thread pool before clean transport caches, there
1975 can be a crash if a thread is executing in transport list loop in
1976 function send_tcp_message.
1977 Also add a check if priv->send_pool in on_message_sent to avoid that a
1978 new thread is pushed during wait of free thread pool. This is possible
1979 since when waiting for free thread pool mutex have to be unlocked.
1981 === release 1.15.90 ===
1983 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
1989 * gst-rtsp-server.doap:
1993 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
1995 * gst/rtsp-server/rtsp-stream.c:
1996 rtsp-stream: Add support for GCM (RFC 7714)
1999 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2001 * gst/rtsp-server/rtsp-session-pool.c:
2002 session pool: fix missing klass-> in klass->create_session
2004 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2007 g-i: pass --quiet to g-ir-scanner
2008 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2009 that we get even if everything works just fine.
2010 We still get g-ir-scanner warnings and compiler warnings if
2011 we pass this option.
2013 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2016 g-i: silence 'nested extern' compiler warnings when building scanner binary
2017 We need a nested extern in our init section for the scanner binary
2018 so we can call gst_init to make sure GStreamer types are initialised
2019 (they are not all lazy init via get_type functions, but some are in
2020 exported variables). There doesn't seem to be any other mechanism to
2021 achieve this, so just remove that warning, it's not important at all.
2023 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2026 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2028 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2030 * gst/rtsp-server/rtsp-media.c:
2031 * tests/check/gst/media.c:
2032 rtsp-media: Handle set state when preparing.
2033 Handle the situation when a call to gst_rtsp_media_set_state is done
2034 when media status is preparing.
2035 Also add unit test for this scenario.
2036 The unit test simulate on a media level when two clients share a (live)
2038 Both clients have done SETUP and got responses. Now client 1 is doing
2039 play and client 2 is just closing the connection.
2040 Then without patch there are a problem when
2041 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2042 And client2 is doing closing connection we can end up in a call
2043 to gst_rtsp_media_set_state when
2044 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2045 shut down media is jumped over .
2046 With this patch and this scenario we wait until
2047 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2048 execute after that and now we will execute the logic for
2051 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2059 === release 1.15.2 ===
2061 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2067 * gst-rtsp-server.doap:
2071 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2073 * gst/rtsp-server/rtsp-media.c:
2074 * tests/check/gst/client.c:
2075 rtsp-media: Fix multicast use case with common media
2084 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2086 * gst/rtsp-server/rtsp-client.c:
2087 * gst/rtsp-server/rtsp-stream.c:
2088 * gst/rtsp-server/rtsp-stream.h:
2089 rtsp-server: remove recursive behavior
2090 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2092 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2094 * gst/rtsp-server/rtsp-client.c:
2095 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2096 And route all messages through the send_func if no send_messages_func
2098 We otherwise break backwards compatibility.
2100 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2102 * docs/libs/gst-rtsp-server-sections.txt:
2103 * gst/rtsp-server/rtsp-client.c:
2104 * gst/rtsp-server/rtsp-client.h:
2105 * gst/rtsp-server/rtsp-stream.c:
2106 rtsp-client: Add support for sending buffer lists directly
2107 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2109 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2111 * docs/libs/gst-rtsp-server-sections.txt:
2112 * gst/rtsp-server/rtsp-client.c:
2113 * gst/rtsp-server/rtsp-media.c:
2114 * gst/rtsp-server/rtsp-stream-transport.c:
2115 * gst/rtsp-server/rtsp-stream-transport.h:
2116 * gst/rtsp-server/rtsp-stream.c:
2117 * gst/rtsp-sink/gstrtspclientsink.c:
2118 rtsp-server: Add support for buffer lists
2119 This adds new functions for passing buffer lists through the different
2120 layers without breaking API/ABI, and enables the appsink to actually
2121 provide buffer lists.
2122 This should already reduce CPU usage and potentially context switches a
2123 bit by passing a whole buffer list from the appsink instead of
2124 individual buffers. As a next step it would be necessary to
2125 a) Add support for a vector of data for the GstRTSPMessage body
2126 b) Add support for sending multiple messages at once to the
2127 GstRTSPWatch and let it be handled internally
2128 c) Adding API to GOutputStream that works like writev()
2129 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2131 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2133 * gst/rtsp-server/rtsp-client.c:
2134 client: Fix crash in close handler
2135 The close handler could trigger a crash because it invalidated the
2136 watch_context while still leaving a source attached to it which would be
2137 cleaned up at a later point.
2139 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2141 * gst/rtsp-server/rtsp-stream.c:
2142 rtsp-stream: Use cached address when allocating sockets
2143 If an address/port was previously decided upon (ex: multicast in the
2144 SDP), then use that instead of re-creating another one
2145 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2147 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2149 * gst/rtsp-server/rtsp-media.c:
2150 rtsp-media: Fix race codition in finish_unprepare
2151 The previous fix for race condition around finish_unprepare where the
2152 function could be called twice assumed that the status wouldn't change
2153 during execution of the function. This assumption is incorrect as the
2154 state may change, for example if an error message arrives from the
2156 Instead a flag keeping track on whether the finish_unprepare function
2157 is currently executing is introduced and checked.
2158 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2160 === release 1.15.1 ===
2162 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2168 * gst-rtsp-server.doap:
2172 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2174 * gst/rtsp-server/rtsp-stream.c:
2175 Add source elements to the pipeline before activation
2176 In plug_src we changed the element state before adding it to
2177 the owner container. This prevented the pipeline from intercepting
2178 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2179 to assign a custom task pool.
2180 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2182 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2185 Automatic update of common submodule
2186 From ed78bee to 59cb678
2188 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2190 * examples/test-appsrc.c:
2191 examples: test-appsrc: fix coding style error
2193 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2195 * examples/test-appsrc.c:
2196 examples: test-appsrc: fix buffer leak
2198 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2200 * gst/rtsp-server/rtsp-media.c:
2201 rtsp-media: Update priv->blocked when linked streams are unblocked.
2202 Media is considered to be blocked when all streams that belong to
2203 that media are blocked.
2204 This patch solves the problem of inconsistent updates of
2205 priv->blocked that are not synchronized with the media state.
2207 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2209 * gst/rtsp-server/rtsp-media.c:
2210 rtsp-media: Don't block streams before seeking
2211 Before the seek operation is performed on media, it's required that
2212 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2213 At this stage, all transport parts (transport sinks) have been successfully
2214 added to the pipeline and there is no need for blocking the streams.
2216 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2218 * tests/check/gst/rtspserver.c:
2219 tests: rtspserver: Add shared media test case for TCP
2221 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2223 * gst/rtsp-server/rtsp-stream.c:
2224 rtsp-stream: Use seqnum-offset for rtpinfo
2225 The sequence number in the rtpinfo is supposed to be the first RTP
2226 sequence number. The "seqnum" property on a payloader is supposed to be
2227 the number from the last processed RTP packet. The sequence number for
2228 payloaders that inherit gstrtpbasepayload will not be correct in case of
2229 buffer lists. In order to fix the seqnum property on the payloaders
2230 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2231 "seqnum-offset" from the "stats" property contains the value of the
2232 very first RTP packet in a stream. The server will, however, try to look
2233 at the last simple in the sink element and only use properties on the
2234 payloader in case there no sink elements yet, and by looking at the last
2235 sample of the sink gives the server full control of which RTP packet it
2236 looks at. If the payloader does not have the "stats" property, "seqnum"
2237 is still used since "seqnum-offset" is only present in as part of
2238 "stats" and this is still an issue not solved with this patch.
2239 Needed for gst-plugins-base!17
2241 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2243 * gst/rtsp-server/rtsp-stream.c:
2244 rtsp-stream: Plug memory leak
2245 Attaching a GSource to a context will increase the refcount. The idle
2246 source will never be free'd since the initial reference is never
2249 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2252 Add Gitlab CI configuration
2253 This commit adds a .gitlab-ci.yml file, which uses a feature
2254 to fetch the config from a centralized repository. The intent is
2255 to have all the gstreamer modules use the same configuration.
2256 The configuration is currently hosted at the gst-ci repository
2257 under the gitlab/ci_template.yml path.
2258 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2260 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2263 * gst-rtsp-server.doap:
2264 Update git locations to gitlab
2266 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2268 * gst/rtsp-server/meson.build:
2269 meson: add new onvif types
2271 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2273 * gst/rtsp-server/meson.build:
2274 Add ONVIF subclass headers to the installed headers in meson.build too
2276 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2278 * gst/rtsp-server/rtsp-server-object.h:
2279 * gst/rtsp-server/rtsp-server.h:
2280 rtsp-server: Declare GstRTSPServer struct before anything else
2281 It's needed by all kinds of other headers, including the ones that are
2282 required for defining the GstRTSPServer struct itself and its API.
2284 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2286 * gst/rtsp-server/rtsp-onvif-client.h:
2287 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2288 * gst/rtsp-server/rtsp-onvif-media.h:
2289 * gst/rtsp-server/rtsp-onvif-server.h:
2290 Mark all ONVIF-specific subclasses as Since 1.14
2292 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2294 * gst/rtsp-server/Makefile.am:
2295 * gst/rtsp-server/meson.build:
2296 * gst/rtsp-server/rtsp-context.h:
2297 * gst/rtsp-server/rtsp-onvif-server.c:
2298 * gst/rtsp-server/rtsp-onvif-server.h:
2299 * gst/rtsp-server/rtsp-server-object.h:
2300 * gst/rtsp-server/rtsp-server-prelude.h:
2301 * gst/rtsp-server/rtsp-server.c:
2302 * gst/rtsp-server/rtsp-server.h:
2303 * gst/rtsp-server/rtsp-session.h:
2304 Include ONVIF types from single-include rtsp-server.h
2305 ... by actually making it a single-include header and moving everything
2306 related to the GstRTSPServer type to rtsp-server-object.h instead.
2307 Otherwise there are too many circular includes.
2308 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2310 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2312 * gst/rtsp-server/rtsp-client.c:
2313 * gst/rtsp-server/rtsp-latency-bin.c:
2314 * gst/rtsp-server/rtsp-stream.c:
2315 * gst/rtsp-server/rtsp-stream.h:
2316 rtsp-stream: use idle source in on_message_sent
2317 When the underlying layers are running on_message_sent, this sometimes
2318 causes the underlying layer to send more data, which will cause the
2319 underlying layer to run callback on_message_sent again. This can go on
2321 To break this chain, we introduce an idle source that takes care of
2322 sending data if there are more to send when running callback
2323 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2325 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2327 * gst/rtsp-server/rtsp-client.c:
2328 rtsp-client: Remove timeout GSource on cleanup
2329 Avoids ending up with races where a timeout would still be around
2330 *after* a client was gone. This could happen rather easily in
2331 RTSP-over-HTTP mode on a local connection, where each RTSP message
2332 would be sent as a different HTTP connection with the same tunnelid.
2333 If not properly removed, that timeout would then try to free again
2334 a client (and its contents).
2336 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2338 * gst/rtsp-server/Makefile.am:
2339 autotools: fix distcheck
2341 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2343 * gst/rtsp-server/Makefile.am:
2344 * gst/rtsp-server/meson.build:
2345 * gst/rtsp-server/rtsp-latency-bin.c:
2346 * gst/rtsp-server/rtsp-latency-bin.h:
2347 * gst/rtsp-server/rtsp-onvif-media.c:
2348 onvif: encapsulate onvif part into a bin
2349 ...and thus do not let onvif affect pipelines latency
2350 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2352 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2354 * tests/check/gst/client.c:
2355 tests: client: Avoid bind() failures in tests
2356 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2358 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2360 * gst/rtsp-server/rtsp-media-factory.c:
2361 * gst/rtsp-server/rtsp-media-factory.h:
2362 * gst/rtsp-server/rtsp-media.c:
2363 * gst/rtsp-server/rtsp-media.h:
2364 * gst/rtsp-server/rtsp-stream.c:
2365 * gst/rtsp-server/rtsp-stream.h:
2366 * tests/check/gst/client.c:
2367 * tests/check/gst/mediafactory.c:
2368 New property for socket binding to mcast addresses
2369 By default the multicast sockets are bound to INADDR_ANY,
2370 as it's not allowed to bind sockets to multicast addresses
2371 in Windows. This default behaviour can be changed by setting
2372 bind-mcast-address property on the media-factory object.
2373 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2375 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2378 * gst/rtsp-server/Makefile.am:
2379 * gst/rtsp-server/meson.build:
2380 * gst/rtsp-server/rtsp-address-pool.c:
2381 * gst/rtsp-server/rtsp-auth.c:
2382 * gst/rtsp-server/rtsp-client.c:
2383 * gst/rtsp-server/rtsp-context.c:
2384 * gst/rtsp-server/rtsp-media-factory-uri.c:
2385 * gst/rtsp-server/rtsp-media-factory.c:
2386 * gst/rtsp-server/rtsp-media.c:
2387 * gst/rtsp-server/rtsp-mount-points.c:
2388 * gst/rtsp-server/rtsp-params.c:
2389 * gst/rtsp-server/rtsp-permissions.c:
2390 * gst/rtsp-server/rtsp-sdp.c:
2391 * gst/rtsp-server/rtsp-server-prelude.h:
2392 * gst/rtsp-server/rtsp-server.c:
2393 * gst/rtsp-server/rtsp-session-media.c:
2394 * gst/rtsp-server/rtsp-session-pool.c:
2395 * gst/rtsp-server/rtsp-session.c:
2396 * gst/rtsp-server/rtsp-stream-transport.c:
2397 * gst/rtsp-server/rtsp-stream.c:
2398 * gst/rtsp-server/rtsp-thread-pool.c:
2399 * gst/rtsp-server/rtsp-token.c:
2401 libs: fix API export/import and 'inconsistent linkage' on MSVC
2402 Export rtsp-server library API in headers when we're building the
2403 library itself, otherwise import the API from the headers.
2404 This fixes linker warnings on Windows when building with MSVC.
2405 Fix up some missing config.h includes when building the lib which
2406 is needed to get the export api define from config.h
2407 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2409 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2411 * gst/rtsp-server/rtsp-media-factory.c:
2412 rtsp-media-factory: Add missing break statements
2413 This resulted in warnings/assertions whenever one accessed the
2414 max-mcast-ttl property.
2418 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2421 * meson_options.txt:
2422 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2424 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2427 * meson_options.txt:
2428 * tests/check/meson.build:
2429 meson: add option to disable build of rtspclientsink plugin
2431 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2433 * meson_options.txt:
2434 meson: re-arrange options
2436 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2439 * meson_options.txt:
2440 * tests/check/meson.build:
2441 * tests/meson.build:
2442 meson: Use feature option for tests option
2443 This was somehow missed the last time around.
2445 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2447 * gst/rtsp-server/meson.build:
2449 meson: Maintain macOS ABI through dylib versioning
2450 Requires Meson 0.48, but the feature will be ignored on older versions
2451 so it's safe to add it without bumping the requirement.
2453 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2455 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2457 * gst/rtsp-sink/meson.build:
2459 meson: add pkg-config file for the rtspclientsink plugin
2461 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2463 * gst/rtsp-server/rtsp-client.c:
2464 * tests/check/gst/client.c:
2465 rtsp-client: Avoid reuse of channel numbers for interleaved
2466 If a (strange) client would reuse interleaved channel numbers in
2467 multiple SETUP requests, we should not accept them. The channel
2468 numbers are used for looking up stream transports in the
2469 priv->transports hash table, and transports disappear from the table
2470 if channel numbers are reused.
2471 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2472 server to change the channel numbers suggested by the client.
2473 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2475 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2477 * tests/check/gst/client.c:
2478 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2479 Allow regex for matching transport header against expected pattern.
2480 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2482 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2484 * tests/check/meson.build:
2485 meson: There is no gstreamer-plugins-good-1.0.pc
2486 There is no installed version of that, only an uninstalled version.
2488 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2490 * gst/rtsp-server/rtsp-client.c:
2491 * tests/check/gst/stream.c:
2492 Fix indentation again
2494 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2496 * gst/rtsp-server/rtsp-client.c:
2497 * gst/rtsp-server/rtsp-stream.c:
2498 * gst/rtsp-server/rtsp-stream.h:
2499 * tests/check/gst/client.c:
2500 * tests/check/gst/stream.c:
2501 stream: Added a list of multicast client addresses
2502 When media is shared, the same media stream can be sent
2503 to multiple multicast groups. Currently, there is no API
2504 to retrieve multicast addresses from the stream.
2505 When calling gst_rtsp_stream_get_multicast_address() function,
2506 only the first multicast address is returned.
2507 With this patch, each multicast destination requested in SETUP
2508 will be stored in an internal list (call to
2509 gst_rtsp_stream_add_multicast_client_address()).
2510 The list of multicast groups requested by the clients can be
2511 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2512 There still exist some problems with the current implementation
2513 in the multicast case:
2514 1) The receiving part is currently only configured with
2515 regard to the first multicast client (see
2516 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2517 2) Secondly, of security reasons, some constraints should be
2518 put on the requested multicast destinations (see
2519 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2520 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2521 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2523 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2525 * gst/rtsp-server/rtsp-client.c:
2526 * gst/rtsp-server/rtsp-stream.c:
2527 * gst/rtsp-server/rtsp-stream.h:
2528 * tests/check/gst/client.c:
2529 stream: Choose the maximum ttl value provided by multicast clients
2530 The maximum ttl value provided so far by the multicast clients
2531 will be chosen and reported in the response to the current
2533 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2534 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2536 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2538 * gst/rtsp-server/rtsp-stream.c:
2539 * tests/check/gst/client.c:
2540 rtsp-stream: Don't require address pool in the transport specific case
2541 If "transport.client-settings" parameter is set to true, the client is
2542 allowed to specify destination, ports and ttl.
2543 There is no need for pre-configured address pool.
2544 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2545 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2547 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2549 * gst/rtsp-server/rtsp-client.c:
2550 * tests/check/gst/client.c:
2551 client: Don't reserve multicast address in the client setting case
2552 When two multicast clients request specific transport
2553 configurations, and "transport.client-settings" parameter is
2554 set to true, it's wrong to actually require that these two
2555 clients request the same multicast group.
2556 Removed test_client_multicast_invalid_transport_specific test
2557 cases as they wrongly require that the requested destination
2558 address is supposed to be present in the address pool, also in
2559 the case when "transport.client-settings" parameter is set to true.
2560 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2561 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2563 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2565 * gst/rtsp-server/rtsp-media-factory.c:
2566 * gst/rtsp-server/rtsp-media-factory.h:
2567 * gst/rtsp-server/rtsp-media.c:
2568 * gst/rtsp-server/rtsp-media.h:
2569 * gst/rtsp-server/rtsp-stream.c:
2570 * gst/rtsp-server/rtsp-stream.h:
2571 * tests/check/gst/mediafactory.c:
2572 Add new API for setting/getting maximum multicast ttl value
2573 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2574 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2576 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2578 * gst/rtsp-server/rtsp-stream.c:
2579 rtsp-stream: avoid duplicating the first multicast client
2580 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2581 clients were dynamically added and removed to the multicast
2582 udp sinks, as such we should no longer add a first client in
2583 set_multicast_socket_for_udpsink
2584 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2586 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2588 * gst/rtsp-server/rtsp-stream.c:
2589 Revert "rtsp-stream: avoid duplicating the first multicast client"
2590 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2591 Commits where accidentially squashed together
2593 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2595 * gst/rtsp-server/rtsp-client.c:
2596 * gst/rtsp-server/rtsp-media-factory.c:
2597 * gst/rtsp-server/rtsp-media-factory.h:
2598 * gst/rtsp-server/rtsp-media.c:
2599 * gst/rtsp-server/rtsp-media.h:
2600 * gst/rtsp-server/rtsp-stream.c:
2601 * gst/rtsp-server/rtsp-stream.h:
2602 * tests/check/gst/client.c:
2603 * tests/check/gst/mediafactory.c:
2604 Revert "Add new API for setting/getting maximum multicast ttl value"
2605 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2606 Commits where accidentially squashed together
2608 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2610 * gst/rtsp-server/rtsp-stream.c:
2611 * tests/check/gst/client.c:
2612 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2613 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2614 Commits where accidentially squashed together
2616 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2618 * gst/rtsp-server/rtsp-client.c:
2619 * gst/rtsp-server/rtsp-stream.c:
2620 * gst/rtsp-server/rtsp-stream.h:
2621 * tests/check/gst/client.c:
2622 * tests/check/gst/stream.c:
2623 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2624 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2625 Commits where accidentially squashed together
2627 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2629 * examples/test-auth-digest.c:
2630 examples: Fix indentation
2632 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2634 * gst/rtsp-server/rtsp-client.c:
2635 * gst/rtsp-server/rtsp-stream.c:
2636 * gst/rtsp-server/rtsp-stream.h:
2637 * tests/check/gst/client.c:
2638 * tests/check/gst/stream.c:
2639 stream: Choose the maximum ttl value provided by multicast clients
2640 The maximum ttl value provided so far by the multicast clients
2641 will be chosen and reported in the response to the current
2643 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2645 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2647 * gst/rtsp-server/rtsp-stream.c:
2648 * tests/check/gst/client.c:
2649 rtsp-stream: Don't require address pool in the transport specific case
2650 If "transport.client-settings" parameter is set to true, the client is
2651 allowed to specify destination, ports and ttl.
2652 There is no need for pre-configured address pool.
2653 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2655 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2657 * gst/rtsp-server/rtsp-client.c:
2658 * gst/rtsp-server/rtsp-media-factory.c:
2659 * gst/rtsp-server/rtsp-media-factory.h:
2660 * gst/rtsp-server/rtsp-media.c:
2661 * gst/rtsp-server/rtsp-media.h:
2662 * gst/rtsp-server/rtsp-stream.c:
2663 * gst/rtsp-server/rtsp-stream.h:
2664 * tests/check/gst/client.c:
2665 * tests/check/gst/mediafactory.c:
2666 Add new API for setting/getting maximum multicast ttl value
2667 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2669 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2671 * gst/rtsp-server/rtsp-stream.c:
2672 rtsp-stream: avoid duplicating the first multicast client
2673 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2674 clients were dynamically added and removed to the multicast
2675 udp sinks, as such we should no longer add a first client in
2676 set_multicast_socket_for_udpsink
2677 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2679 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2681 * gst/rtsp-server/Makefile.am:
2682 rtsp-server: Add gstreamer-base gir dir in autotools
2684 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2686 * gst/rtsp-server/rtsp-client.c:
2687 * gst/rtsp-server/rtsp-stream.c:
2688 rtsp-client: always allocate both IPV4 and IPV6 sockets
2689 multiudpsink does not support setting the socket* properties
2690 after it has started, which meant that rtsp-server could no
2691 longer serve on both IPV4 and IPV6 sockets since the patches
2692 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2694 When first connecting an IPV6 client then an IPV4 client,
2695 multiudpsink fell back to using the IPV6 socket.
2696 When first connecting an IPV4 client, then an IPV6 client,
2697 multiudpsink errored out, released the IPV4 socket, then
2698 crashed when trying to send a message on NULL nevertheless,
2699 that is however a separate issue.
2700 This could probably be fixed by handling the setting of
2701 sockets in multiudpsink after it has started, that will
2702 however be a much more significant effort.
2703 For now, this commit simply partially reverts the behaviour
2704 of rtsp-stream: it will continue to only create the udpsinks
2705 when needed, as was the case since the patches were merged,
2706 it will however when creating them, always allocate both
2707 sockets and set them on the sink before it starts, as was
2708 the case prior to the patches.
2709 Transport configuration will only error out if the allocation
2710 of UDP sockets fails for the actual client's family, this
2711 also downgrades the GST_ERRORs in alloc_ports_one_family
2712 to GST_WARNINGs, as failing to allocate is no longer
2714 https://bugzilla.gnome.org/show_bug.cgi?id=796875
2716 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2719 * meson_options.txt:
2720 meson: Convert common options to feature options
2721 These are necessary for gst-build to set options correctly. The
2722 remaining automagic option is cgroup support in examples.
2723 https://bugzilla.gnome.org/show_bug.cgi?id=795107
2725 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2727 * gst/rtsp-server/rtsp-stream.c:
2728 rtsp-stream: Slightly simplify locking
2730 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
2732 * gst/rtsp-server/rtsp-client.c:
2733 * gst/rtsp-server/rtsp-stream-transport.c:
2734 * gst/rtsp-server/rtsp-stream-transport.h:
2735 * gst/rtsp-server/rtsp-stream.c:
2736 Limit queued TCP data messages to one per stream
2737 Before, the watch backlog size in GstRTSPClient was changed
2738 dynamically between unlimited and a fixed size, trying to avoid both
2739 unlimited memory usage and deadlocks while waiting for place in the
2740 queue. (Some of the deadlocks were described in a long comment in
2742 In the previous commit, we changed to a fixed backlog size of 100.
2743 This is possible, because we now handle RTP/RTCP data messages differently
2744 from RTSP request/response messages.
2745 The data messages are messages tunneled over TCP. We allow at most one
2746 queued data message per stream in GstRTSPClient at a time, and
2747 successfully sent data messages are acked by sending a "message-sent"
2748 callback from the GstStreamTransport. Until that ack comes, the
2749 GstRTSPStream does not call pull_sample() on its appsink, and
2750 therefore the streaming thread in the pipeline will not be blocked
2751 inside GstRTSPClient, waiting for a place in the queue.
2752 pull_sample() is called when we have both an ack and a "new-sample"
2753 signal from the appsink. Then, we know there is a buffer to write.
2754 RTSP request/response messages are not acked in the same way as data
2755 messages. The rest of the 100 places in the queue are used for
2756 them. If the queue becomes full of request/response messages, we
2757 return an error and close the connection to the client.
2758 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2760 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
2762 * gst/rtsp-server/rtsp-client.c:
2763 rtsp-client: Use fixed backlog size
2764 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
2765 Preparation for the next commit, which changes to a different way of
2766 avoiding both deadlocks and unlimited memory usage with the watch
2769 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2771 * gst/rtsp-server/rtsp-media.c:
2772 rtsp-media: unref clock (if set) when finalizing
2773 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2775 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2777 * docs/libs/gst-rtsp-server-sections.txt:
2778 rtsp-media: add gst_rtsp_media_*_set_clock to docs
2779 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2781 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2783 * gst/rtsp-server/rtsp-media-factory.c:
2784 media-factory: unref old clock when setting new clock
2785 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2787 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
2789 * gst/rtsp-server/rtsp-media-factory.c:
2790 media-factory: unref clock in finalize
2791 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2793 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2795 * gst/rtsp-server/rtsp-onvif-media.c:
2796 rtsp-onvif-media: fix g-ir-scanner warnings
2798 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2801 .gitignore: add another example binary
2803 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
2805 * examples/meson.build:
2806 meson: add new test-appsrc2 example to meson build
2808 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2810 * examples/Makefile.am:
2811 examples: fix build of new test-appsrc2 example
2812 Need to link against libgstapp-1.0.
2814 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
2816 * examples/.gitignore:
2817 * examples/Makefile.am:
2818 * examples/test-appsrc2.c:
2819 examples: Add test-appsrc2
2820 Add an example of feeding both audio and video into an RTSP
2821 pipeline via appsrc.
2823 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
2825 * gst/rtsp-server/rtsp-client.c:
2826 client: Strip transport parts as whitespaces could be around commas
2827 https://bugzilla.gnome.org/show_bug.cgi?id=758428
2829 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
2831 * gst/rtsp-server/rtsp-stream.c:
2832 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
2833 Fix race when setting up source elements.
2834 Since we set the source element(s) to PLAYING state before hooking
2835 them up to the downstream funnel, it's possible for the source element
2836 to receive packets before we actually get to linking it to the funnel,
2837 in which case buffers would be pushed out on an unlinked pad, causing
2838 it to error out and stop receiving more data.
2839 We fix this by blocking the source's srcpad until we have linked it.
2840 https://bugzilla.gnome.org/show_bug.cgi?id=796160
2842 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
2844 * gst/rtsp-server/rtsp-stream.c:
2845 rtsp-stream: Fix mismatch between allowed and configured protocols
2846 https://bugzilla.gnome.org/show_bug.cgi?id=796679
2848 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
2850 * gst/rtsp-server/rtsp-stream.c:
2851 rtsp-stream: Emit a signal when the SRTP decoder is created
2852 https://bugzilla.gnome.org/show_bug.cgi?id=778080
2854 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
2856 * gst/rtsp-server/rtsp-stream.c:
2857 rtsp-stream: Don't require presence of sinks in _get_*_socket()
2858 Transport specific sink elements are added to the pipeline
2859 in PLAY request and sockets are already created in SETUP so
2860 it's actually wrong to require the presence of sinks in
2861 _get_*_socket() functions.
2862 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2864 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
2866 * gst/rtsp-server/rtsp-stream.c:
2867 rtsp-stream: Update transport for multicast clients as well
2868 If a multicast client requests different transport settings
2869 than the existing one make sure that this new transport
2870 configuruation is propagated to the multicast udp sink.
2871 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2873 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
2875 * gst/rtsp-server/rtsp-stream.c:
2876 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
2877 And not on unicast udp sinks
2878 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2880 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
2882 * gst/rtsp-server/rtsp-address-pool.c:
2883 * gst/rtsp-server/rtsp-auth.c:
2884 * gst/rtsp-server/rtsp-client.c:
2885 * gst/rtsp-server/rtsp-media-factory-uri.c:
2886 * gst/rtsp-server/rtsp-media-factory.c:
2887 * gst/rtsp-server/rtsp-media.c:
2888 * gst/rtsp-server/rtsp-mount-points.c:
2889 * gst/rtsp-server/rtsp-server.c:
2890 * gst/rtsp-server/rtsp-session-media.c:
2891 * gst/rtsp-server/rtsp-session-pool.c:
2892 * gst/rtsp-server/rtsp-session.c:
2893 * gst/rtsp-server/rtsp-stream-transport.c:
2894 * gst/rtsp-server/rtsp-stream.c:
2895 * gst/rtsp-server/rtsp-thread-pool.c:
2896 Update for g_type_class_add_private() deprecation in recent GLib
2898 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
2900 * gst/rtsp-server/rtsp-auth.c:
2901 * gst/rtsp-server/rtsp-media.c:
2902 * gst/rtsp-server/rtsp-sdp.c:
2903 * gst/rtsp-server/rtsp-stream.c:
2906 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
2908 * examples/Makefile.am:
2909 * examples/test-video-disconnect.c:
2910 examples: Add test-video-disconnect example
2911 Simple example which cuts off all clients 10 seconds
2912 after the first one connects.
2914 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2916 * docs/libs/gst-rtsp-server-sections.txt:
2917 * examples/test-auth-digest.c:
2918 * gst/rtsp-server/rtsp-auth.c:
2919 * gst/rtsp-server/rtsp-auth.h:
2920 rtsp-auth: Add support for parsing .htdigest files
2921 Passwords are usually not stored in clear text, but instead
2922 stored already hashed in a .htdigest file.
2923 Add support for parsing such files, add API to allow setting
2924 a custom realm in RTSPAuth, and update the digest example.
2925 https://bugzilla.gnome.org/show_bug.cgi?id=796637
2927 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
2929 * gst/rtsp-sink/gstrtspclientsink.c:
2930 * gst/rtsp-sink/gstrtspclientsink.h:
2931 rtspclientsink: fix waiting for multiple streams
2932 We were previously only ever waiting for a single stream to notify it's
2933 blocked status through GstRTSPStreamBlocking. Actually count streams to
2935 Fixes rtspclientsink sending SDP's without out some of the input
2937 https://bugzilla.gnome.org/show_bug.cgi?id=796624
2939 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2941 * docs/libs/gst-rtsp-server-sections.txt:
2942 docs: add missing auth methods
2944 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2946 * gst/rtsp-server/rtsp-stream.c:
2947 rtsp-stream: only create funnel if it didn't exist already.
2948 This precented using multiple protocols for the same stream.
2949 https://bugzilla.gnome.org/show_bug.cgi?id=796634
2951 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2953 * examples/meson.build:
2954 meson: build auth-digest example
2956 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
2958 * gst/rtsp-server/rtsp-client.c:
2959 * gst/rtsp-server/rtsp-media.c:
2960 * gst/rtsp-server/rtsp-sdp.c:
2961 * gst/rtsp-server/rtsp-session-media.c:
2962 * gst/rtsp-server/rtsp-stream-transport.c:
2963 Get payloader stats only for the sending streams
2964 Get/set payloader properties only for streams that actually
2965 contain a payloader element.
2966 https://bugzilla.gnome.org/show_bug.cgi?id=796523
2968 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
2970 * gst/rtsp-server/Makefile.am:
2971 Makefile: Don't hardcode libtool for g-i build
2972 Similar to the other commits in core/base/bad
2974 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
2976 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2977 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
2978 https://bugzilla.gnome.org/show_bug.cgi?id=796229
2980 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
2982 * gst/rtsp-sink/gstrtspclientsink.c:
2983 rtspclientsink: Don't deadlock in preroll on early close
2984 If the connection is closed very early, the flushing
2985 marker might not get set and rtspclientsink can get
2986 deadlocked waiting for preroll forever.
2987 https://bugzilla.gnome.org/show_bug.cgi?id=786961
2989 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2992 * meson_options.txt:
2993 meson: Update option names to omit disable_ and with- prefixes
2994 Also yield common options to the outer project (gst-build in our case)
2995 so that they don't have to be set manually.
2997 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3000 meson: use -Wl,-Bsymbolic-functions where supported
3001 Just like the autotools build.
3003 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3006 * tests/check/Makefile.am:
3007 configure: check for -good and -bad plugins only in uninstalled setup
3008 Avoids confusing configure messages looking or a -good .pc file
3010 Also use plugindir variables that common macros set while at it.
3011 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3013 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3015 * gst/rtsp-server/rtsp-client.c:
3016 rtsp-client: Fix session timeout
3017 When streaming data over TCP then is not the keep-alive
3018 functionality working.
3019 The reason is that the function do_send_data have changed
3020 to boolean but the code is still checking the received result
3021 from send_func with GST_RTSP_OK.
3022 The result is that a successful send_func will always lead to
3023 that do_send_data is returning false and the keep-alive will
3025 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3027 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3029 * docs/libs/gst-rtsp-server-sections.txt:
3030 * gst/rtsp-server/rtsp-media.c:
3031 * gst/rtsp-server/rtsp-sdp.c:
3032 * gst/rtsp-server/rtsp-stream.c:
3033 * gst/rtsp-server/rtsp-stream.h:
3034 * gst/rtsp-sink/gstrtspclientsink.c:
3035 * gst/rtsp-sink/gstrtspclientsink.h:
3036 Implement support for ULP Forward Error Correction
3037 In this initial commit, interface is only exposed for RECORD,
3038 further work will be needed in rtspsrc to support this for
3040 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3042 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3044 * gst/rtsp-server/rtsp-onvif-media.c:
3045 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3046 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3047 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3048 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3049 the opposite, just like the ONVIF standard.
3050 Let's follow those RFCs as we're doing RTSP here, and add a property at
3051 a later time if needed to switch to the SDP RFC behaviour.
3052 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3054 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3057 Automatic update of common submodule
3058 From 3fa2c9e to ed78bee
3060 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3062 * gst/rtsp-server/rtsp-client.c:
3063 * gst/rtsp-server/rtsp-media-factory.c:
3064 * gst/rtsp-server/rtsp-media.c:
3065 * gst/rtsp-server/rtsp-stream.c:
3066 * tests/check/gst/rtspclientsink.c:
3067 gst: Run everything through gst-indent again
3069 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3071 * gst/rtsp-server/rtsp-media.c:
3072 * tests/check/gst/media.c:
3073 rtsp-media: query the position on active streams if media is complete
3074 If the media is complete, i.e. one or more streams have been configured
3075 with sinks, then we want to query the position on those streams only.
3076 A query on an incomplete stream may return a position that originates from
3078 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3080 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3082 * gst/rtsp-sink/gstrtspclientsink.c:
3083 rtspclientsink: make sure not to use freed string
3084 Set transport string to NULL after freeing it, so that
3085 at worst we get a NULL pointer if constructing a new
3086 transport string fails (which shouldn't really fail here).
3087 Also check return value of that, just in case.
3090 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3092 * gst/rtsp-server/rtsp-client.c:
3093 rtsp-client: do not free string passed to take_header
3095 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3097 * gst/rtsp-server/rtsp-stream.c:
3098 rtsp-stream: do not take lock in request_aux_receiver
3099 Added it right before pushing the previous commit, it is
3100 incorrect and deadlocks because this function gets called
3101 from the join_bin thread, which already holds the lock,
3102 that's the reason why request_aux_sender didn't take the
3105 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3107 * docs/libs/gst-rtsp-server-sections.txt:
3108 * gst/rtsp-server/rtsp-media-factory.c:
3109 * gst/rtsp-server/rtsp-media-factory.h:
3110 * gst/rtsp-server/rtsp-media.c:
3111 * gst/rtsp-server/rtsp-media.h:
3112 * gst/rtsp-server/rtsp-stream.c:
3113 * gst/rtsp-server/rtsp-stream.h:
3114 rtsp-server: add API to enable retransmission requests
3115 "do-retransmission" was previously set when rtx-time != 0,
3116 which made no sense as do-retransmission is used to enable
3117 the sending of retransmission requests, where as rtx-time
3118 is used by the peer to enable storing of buffers in order
3119 to respond to retransmission requests.
3120 rtsp-media now also provides a callback for the
3121 request-aux-receiver signal.
3122 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3124 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3126 * gst/rtsp-sink/gstrtspclientsink.c:
3127 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3128 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3130 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3132 * gst/rtsp-sink/gstrtspclientsink.c:
3133 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3134 This in order to be able to decrypt the RTCP backchannel
3135 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3137 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3139 * gst/rtsp-server/rtsp-client.c:
3140 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3141 When sending back an encrypted RTCP back channel, it is useful
3142 for the client to know the encryption key.
3143 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3145 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3147 * gst/rtsp-server/rtsp-client.c:
3148 * gst/rtsp-server/rtsp-stream.c:
3149 * gst/rtsp-server/rtsp-stream.h:
3150 rtsp-stream: extract handle_keymgmt from rtsp-client
3151 rtspclientsink will also need to parse KeyMgmt headers
3152 sent by the server to decrypt the RTCP backchannel stream
3153 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3155 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3157 * gst/rtsp-sink/gstrtspclientsink.c:
3158 * tests/check/gst/rtspclientsink.c:
3159 rtspclientsink: Fix client ports for the RTCP backchannel
3160 This was broken since the work for delayed transport creation
3161 was merged: the creation of the transports string depends on
3162 calling stream_get_server_port, which only starts returning
3163 something meaningful after a call to stream_allocate_udp_sockets
3164 has been made, this function expects a transport that we parse
3165 from the transport string ...
3166 Significant refactoring is in order, but does not look entirely
3167 trivial, for now we put a band aid on and create a second transport
3168 string after the stream has been completed, to pass it in
3169 the request headers instead of the previous, incomplete one.
3170 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3172 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3174 * gst/rtsp-server/rtsp-client.c:
3175 rtsp-client:Error handling when equal http session cookie
3176 There are some clients that are sending same session cookie on random
3178 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3180 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3182 * gst/rtsp-server/rtsp-media-factory-uri.c:
3183 rtsp-media-factory-uri: Fix compilation with latest GLib
3184 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3185 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3186 data->factory = g_object_ref (factory);
3189 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3197 === release 1.14.0 ===
3199 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3205 * gst-rtsp-server.doap:
3209 === release 1.13.91 ===
3211 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3217 * gst-rtsp-server.doap:
3221 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3223 * gst/rtsp-server/Makefile.am:
3224 * gst/rtsp-server/meson.build:
3225 * gst/rtsp-server/rtsp-address-pool.h:
3226 * gst/rtsp-server/rtsp-auth.h:
3227 * gst/rtsp-server/rtsp-client.h:
3228 * gst/rtsp-server/rtsp-context.h:
3229 * gst/rtsp-server/rtsp-media-factory-uri.h:
3230 * gst/rtsp-server/rtsp-media-factory.h:
3231 * gst/rtsp-server/rtsp-media.h:
3232 * gst/rtsp-server/rtsp-mount-points.h:
3233 * gst/rtsp-server/rtsp-onvif-client.h:
3234 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3235 * gst/rtsp-server/rtsp-onvif-media.h:
3236 * gst/rtsp-server/rtsp-onvif-server.h:
3237 * gst/rtsp-server/rtsp-params.h:
3238 * gst/rtsp-server/rtsp-permissions.h:
3239 * gst/rtsp-server/rtsp-sdp.h:
3240 * gst/rtsp-server/rtsp-server-prelude.h:
3241 * gst/rtsp-server/rtsp-server.h:
3242 * gst/rtsp-server/rtsp-session-media.h:
3243 * gst/rtsp-server/rtsp-session-pool.h:
3244 * gst/rtsp-server/rtsp-session.h:
3245 * gst/rtsp-server/rtsp-stream-transport.h:
3246 * gst/rtsp-server/rtsp-stream.h:
3247 * gst/rtsp-server/rtsp-thread-pool.h:
3248 * gst/rtsp-server/rtsp-token.h:
3249 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3250 We need different export decorators for the different libs.
3251 For now no actual change though, just rename before the release,
3252 and add prelude headers to define the new decorator to GST_EXPORT.
3254 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3256 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3257 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3258 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3260 === release 1.13.90 ===
3262 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3268 * gst-rtsp-server.doap:
3272 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3274 * gst/rtsp-server/rtsp-media-factory.c:
3275 * gst/rtsp-server/rtsp-permissions.c:
3276 permissions: add Since tags and example for new API
3278 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3280 * docs/libs/gst-rtsp-server-sections.txt:
3281 * gst/rtsp-server/rtsp-media-factory.c:
3282 * gst/rtsp-server/rtsp-media-factory.h:
3283 * gst/rtsp-server/rtsp-permissions.c:
3284 * gst/rtsp-server/rtsp-permissions.h:
3285 * tests/check/gst/permissions.c:
3286 permissions: more bindings-friendly API
3287 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3289 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3292 meson: enable more warnings
3294 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3296 * gst/rtsp-server/rtsp-client.c:
3297 rtsp-client: Place netaddress meta on packets received via TCP
3298 This allows us to later map signals from rtpbin/rtpsource back to the
3299 corresponding stream transport, and allows to do keep-alive based on
3300 RTCP packets in case of TCP media transport.
3301 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3303 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3305 * gst/rtsp-sink/gstrtspclientsink.c:
3306 rtspclientsink: if OPEN failed, unqueue next command
3307 As READY_TO_PAUSED can no longer return async, the RECORD
3308 command will be queued before the OPEN command fails
3309 (for example in case the server could not be connected),
3310 and record then waits for ever.
3311 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3313 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3315 * gst/rtsp-sink/gstrtspclientsink.c:
3316 rtspclientsink: fix retrieval of custom payloader caps
3317 If a bin is passed as the custom payloader, the caps of
3318 its factory will be empty, the correct way to obtain the caps
3319 is to query its sinkpad.
3321 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3323 * gst/rtsp-sink/gstrtspclientsink.c:
3324 rtspclientsink: fix extra unref of custom payloader
3326 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3328 * gst/rtsp-sink/gstrtspclientsink.c:
3329 rspclientsink: fix recent code indentation
3331 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3333 * gst/rtsp-sink/gstrtspclientsink.c:
3334 rtspclientsink: add missing get_type prototype
3336 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3338 * gst/rtsp-sink/gstrtspclientsink.c:
3339 rtspclientsink: allow setting payloader as pad property
3340 This was a FIXME item, and can be quite useful, also
3341 allowing to specify payloader properties from the command
3342 line, which is always nice.
3343 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3345 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3347 * gst/rtsp-server/rtsp-media.c:
3348 rtsp-media: Replace g_print() log line
3349 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3351 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3353 * gst/rtsp-server/rtsp-media.c:
3354 * tests/check/gst/rtspclientsink.c:
3355 rtsp-media: fix RECORD getting stuck
3356 The test_record case was working because async=false had
3357 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3358 but that was incorrect, as it should not be needed.
3359 Removing async=false made the test fail as expected, this is
3360 fixed by not trying to preroll when preparing the media for
3361 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3362 and our peer will not start sending media until it has received
3363 a response to that request, and sent and received a response
3364 to RECORD as well, thus obviously preventing preroll.
3365 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3367 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3369 * gst/rtsp-server/rtsp-auth.c:
3370 rtsp-auth: fix set_tls_authentication_mode annotation
3372 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3374 * gst/rtsp-server/rtsp-onvif-media.c:
3375 rtp-server: remove redefined variable
3376 res is a boolean variable which is defined in the function scope and
3377 redefined, with no reason, in the loop scope. This patch removes the
3379 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3381 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3383 * gst/rtsp-server/rtsp-media.c:
3384 * gst/rtsp-server/rtsp-stream.c:
3385 * gst/rtsp-server/rtsp-stream.h:
3386 stream: Add functions for checking if stream is receiver or sender
3387 ...and replace all checks for RECORD in GstRTSPMedia which are really
3388 for "sender-only". This way the code becomes more generic and introducing
3389 support for onvif-backchannel later on will require no changes in
3392 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3394 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3395 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3396 onvif: Make requires_backchannel() public
3397 ...in order to let subclasses building the onvif part of the pipeline
3398 check whether backchannel shall be included or not.
3400 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3402 * gst/rtsp-server/rtsp-onvif-media.c:
3403 rtsp-server: Switch around sendonly/recvonly attributes
3404 They are wrong in the ONVIF streaming spec. The backchannel should be
3405 recvonly and the normal media should be sendonly: direction is always
3406 from the point of view of the SDP offerer (the server) according to
3409 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3411 * docs/libs/gst-rtsp-server-docs.sgml:
3412 * docs/libs/gst-rtsp-server-sections.txt:
3413 * examples/.gitignore:
3414 * examples/Makefile.am:
3415 * examples/test-onvif-backchannel.c:
3416 * gst/rtsp-server/Makefile.am:
3417 * gst/rtsp-server/rtsp-media.h:
3418 * gst/rtsp-server/rtsp-onvif-client.c:
3419 * gst/rtsp-server/rtsp-onvif-client.h:
3420 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3421 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3422 * gst/rtsp-server/rtsp-onvif-media.c:
3423 * gst/rtsp-server/rtsp-onvif-media.h:
3424 * gst/rtsp-server/rtsp-onvif-server.c:
3425 * gst/rtsp-server/rtsp-onvif-server.h:
3426 * gst/rtsp-server/rtsp-sdp.c:
3427 * gst/rtsp-server/rtsp-sdp.h:
3428 rtsp: Add support for ONVIF backchannel
3429 This adds a new RTSP server, client, media-factory and media subclass
3430 for handling the specifics of the backchannel. Ideally this later can be
3431 extended with other ONVIF specific features.
3433 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3435 * gst/rtsp-server/rtsp-media.c:
3436 rtsp-media: Add support for sending+receiving medias
3437 We need to add an appsrc/appsink in that case because otherwise the
3438 media bin will be a sink and a source for rtpbin, causing a pipeline
3440 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3442 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3448 === release 1.13.1 ===
3450 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3454 * gst-rtsp-server.doap:
3458 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3460 * gst/rtsp-server/rtsp-session-pool.c:
3461 session-pool: remove nullable return annotation
3462 create_watch can only return NULL from the API guards, no
3465 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3467 * gst/rtsp-server/rtsp-media-factory.c:
3468 * gst/rtsp-server/rtsp-media.c:
3469 set_clock functions: Add nullable annotations
3471 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3473 * gst/rtsp-server/rtsp-auth.c:
3474 * gst/rtsp-server/rtsp-client.c:
3475 * gst/rtsp-server/rtsp-media-factory.c:
3476 * gst/rtsp-server/rtsp-media.c:
3477 * gst/rtsp-server/rtsp-mount-points.c:
3478 * gst/rtsp-server/rtsp-server.c:
3479 * gst/rtsp-server/rtsp-session-media.c:
3480 * gst/rtsp-server/rtsp-session-pool.c:
3481 * gst/rtsp-server/rtsp-session.c:
3482 * gst/rtsp-server/rtsp-stream-transport.c:
3483 * gst/rtsp-server/rtsp-stream.c:
3484 * gst/rtsp-server/rtsp-thread-pool.c:
3485 All around: add annotations and API guards
3487 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3489 * tests/test-cleanup.c:
3490 test-cleanup: bind any port
3491 The meson test suite runs tests in parallel, trying to bind
3492 a single port made the test fail.
3494 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3497 meson: make version numbers ints and fix int/string comparison
3498 WARNING: Trying to compare values of different types (str, int).
3499 The result of this is undefined and will become a hard error
3500 in a future Meson release.
3502 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3504 * gst/rtsp-server/rtsp-context.c:
3505 gst_rtsp_context_get_current: add (skip) annotation
3506 The return value type is defined with G_DEFINE_POINTER_TYPE,
3507 and gi emits the following warning:
3508 Invalid non-constant return of bare structure or union; register as
3509 boxed type or (skip)
3511 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3513 * gst/rtsp-server/rtsp-client.c:
3514 rtsp-client: add type annotations
3515 gi doesn't seem to be able to figure out the type of the
3516 signal parameters when defined with G_DEFINE_POINTER_TYPE
3518 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3521 autotools: use -fno-strict-aliasing where supported
3522 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3524 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3527 meson: use -fno-strict-aliasing where supported
3528 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3530 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3532 * gst/rtsp-server/rtsp-mount-points.c:
3533 mount-points: bail out of loop again when matching mount points
3534 Previous patch led to us iterating the entire sequence. Bail out
3535 of the loop again if we have a match but are moving away from it.
3536 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3538 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3540 * tests/check/gst/mountpoints.c:
3541 tests: mountpoints: add more checks for mount point path matching
3542 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3544 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3546 * gst/rtsp-server/rtsp-mount-points.c:
3547 mount-points: fix matching of paths where there's also an entry with a common prefix
3548 e.g. with the following mount points
3552 _match() would not match /raw/video and /raw/snapshot correctly.
3553 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3555 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3557 * docs/libs/gst-rtsp-server-sections.txt:
3558 * gst/rtsp-server/rtsp-permissions.c:
3559 * gst/rtsp-server/rtsp-permissions.h:
3560 * tests/check/gst/permissions.c:
3561 permissions: add some new API to make this usable from bindings
3562 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3564 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3566 * gst/rtsp-server/rtsp-token.c:
3567 rtsp-token: annotate constructors for bindings
3568 This maps _new_empty() to _new(), which also makes RTSPToken()
3569 work properly now. Since this API wasn't usable from bindings
3570 before, this should hopefully be fine.
3571 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3573 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3575 * docs/libs/gst-rtsp-server-sections.txt:
3576 * gst/rtsp-server/rtsp-token.c:
3577 * gst/rtsp-server/rtsp-token.h:
3578 * tests/check/gst/token.c:
3579 rtsp-token: add some API to set fields from bindings
3580 The existing functions are all vararg-based and as such
3581 not usable from bindings.
3582 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3584 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3586 * tests/check/gst/rtspclientsink.c:
3587 * tests/check/gst/rtspserver.c:
3588 * tests/check/gst/sessionpool.c:
3589 * tests/check/gst/stream.c:
3590 tests: fix indentation
3593 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3595 * tests/check/gst/rtspserver.c:
3596 tests: rtspserver: fix another ref leak
3597 Even if this didn't show up in valgrind.
3599 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3601 * tests/check/gst/rtspclientsink.c:
3602 tests: rtspclientsink: fix leak
3604 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3606 * tests/check/gst/rtspserver.c:
3607 test: rtspserver: plug memory leak in test_no_session_timeout
3608 In test_no_session_timeout, unref the rtsp session object when the
3610 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3612 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3614 * gst/rtsp-sink/gstrtspclientsink.c:
3615 rtpsclientsink: Initialize and clear newly added mutex and cond
3616 While it *did* work, glib would automatically create new mutex and cond
3617 ... which never got freed
3619 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3621 * gst/rtsp-server/rtsp-stream.c:
3622 rtsp-stream: Set multicast TTL on the multicast sockets
3623 And not if we do unicast UDP.
3624 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3626 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3628 * gst/rtsp-server/rtsp-stream.c:
3629 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3630 In the multicast case (as in test-multicast, not test-multicast2), the
3631 address could be allocated/reserved (and thus set) already without
3632 allocating the actual socket. We need to allocate the socket here still
3633 instead of just claiming that it was already allocated.
3634 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3636 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3638 * gst/rtsp-sink/gstrtspclientsink.c:
3639 * gst/rtsp-sink/gstrtspclientsink.h:
3640 rtspclientsink: Use the new rtsp-stream API
3641 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3643 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3645 * gst/rtsp-sink/gstrtspclientsink.c:
3646 * gst/rtsp-sink/gstrtspclientsink.h:
3647 rtspclientsink: Wait until OPEN has been scheduled
3648 Make sure that the sink thread has started opening connection
3649 to the server before continuing.
3650 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3652 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3655 Automatic update of common submodule
3656 From e8c7a71 to 3fa2c9e
3658 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3660 * gst/rtsp-server/rtsp-media.c:
3661 * gst/rtsp-server/rtsp-session-media.c:
3662 * gst/rtsp-server/rtsp-stream.c:
3663 rtsp-server: Minor doc fixes
3666 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3669 * tests/Makefile.am:
3670 tests: disable all tests when --disable-tests is used
3671 Move conditional subdir include into top level.
3672 Based on patch by: Joel Holdsworth
3673 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3675 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3678 * meson_options.txt:
3679 * tests/meson.build:
3680 meson: build more tests and add options to disable tests and examples
3682 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3684 * gst/rtsp-server/rtsp-session.c:
3685 Fix build when -Werror=deprecated-declarations is on
3686 As gst_rtsp_session_next_timeout is deprecated.
3688 ../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]
3689 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3691 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3692 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3693 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3696 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3699 Automatic update of common submodule
3700 From 3f4aa96 to e8c7a71
3702 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3704 * tests/check/gst/media.c:
3705 check/media: Add seekability test case: not all streams are active
3706 Media contains two streams but only one is complete and prepared
3708 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3710 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3712 * gst/rtsp-server/rtsp-stream.c:
3713 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3714 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3716 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3718 * gst/rtsp-server/rtsp-media.c:
3719 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
3720 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3722 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
3725 meson: remove vs_module_defs_dir variable which is no longer needed
3727 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
3729 * gst/rtsp-server/rtsp-session.h:
3732 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
3735 * gst/rtsp-server/meson.build:
3737 * win32/common/libgstrtspserver.def:
3738 win32: remove .def file with exports
3739 They're no longer needed, symbol exporting is now explicit
3740 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
3742 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3745 autotools: stop controlling symbol visibility with -export-symbols-regex
3746 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
3747 This should result in consistent behaviour for the autotools and
3750 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
3752 * gst/rtsp-server/rtsp-media.h:
3753 * gst/rtsp-server/rtsp-server.h:
3754 * gst/rtsp-server/rtsp-session.c:
3755 * gst/rtsp-server/rtsp-session.h:
3756 rtsp-server: add missing GST_EXPORT and export deprecated funcs
3758 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
3760 * tests/check/gst/media.c:
3761 check: Add seekability testing on medias
3762 Make sure that once GstRTSPMedia are prepared they returned
3763 the expected seekability results
3764 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3766 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
3768 * docs/libs/gst-rtsp-server-sections.txt:
3769 * gst/rtsp-server/rtsp-media.c:
3770 * gst/rtsp-server/rtsp-stream.c:
3771 * gst/rtsp-server/rtsp-stream.h:
3772 * win32/common/libgstrtspserver.def:
3773 rtsp-media: Enable seeking query before pipeline is complete
3774 SDP are now provided *before* the pipeline is fully complete. In order
3775 to know whether a media is seekable or not therefore requires asking
3776 the invididual streams.
3777 API: gst_rtsp_stream_seekable
3778 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3780 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
3782 * gst/rtsp-server/rtsp-media.c:
3783 rtsp-media: Fix handling in default_unsuspend()
3784 Handle the case when streams are not blocked and media
3785 is suspended from PAUSED.
3786 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
3787 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3789 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
3791 * tests/check/gst/media.c:
3792 check/media: Fix thread pool leak.
3793 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
3794 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3796 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
3798 * gst/rtsp-server/rtsp-media.c:
3799 rtsp-media: Removed fakesink elements
3800 There is not need of adding fakesink elements to the media
3801 pipeline in the dynamic-payloader case.
3802 The media pipeline itself is dynamically updated with
3803 the receiver and sender parts that are based on the client
3804 transport information known after SETUP has been received.
3805 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
3806 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3808 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
3810 * gst/rtsp-server/rtsp-media.c:
3811 rtsp-media: Corrected ASYNC_DONE handling
3812 Media is complete when all the transport based parts are
3813 added to the media pipeline. At this point ASYNC_DONE is
3814 posted by the media pipeline and media is ready to enter
3816 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
3817 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3819 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
3821 * tests/check/gst/media.c:
3822 check/media: Check that prepared media can provide a SDP
3823 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
3825 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
3827 * gst/rtsp-server/rtsp-client.c:
3828 rtsp-client: Don't leak addr
3831 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
3833 * gst/rtsp-server/rtsp-client.c:
3834 * gst/rtsp-server/rtsp-session-media.c:
3835 * gst/rtsp-server/rtsp-stream.c:
3838 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
3840 * gst/rtsp-server/rtsp-media.c:
3841 rtsp-media: Don't unblock with remaining dynamic payloaders
3842 If we still have some dynamic paylaoders which haven't posted
3843 no-more-pads yet, don't go to PREPARED if one of the streams
3845 The risk was that we would end up not exposing/using all specified
3847 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
3848 then it will take a bit more time to start. But only if those 3
3849 conditions are present.
3850 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3852 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
3854 * gst/rtsp-server/rtsp-media.c:
3857 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
3859 * gst/rtsp-server/rtsp-media.c:
3860 rtsp-media: Don't set float on a gint64 variable
3861 Just use 0. Fixes 'undefined' behaviour from clang
3863 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
3865 * gst/rtsp-server/rtsp-media.c:
3866 rtsp-media: Fix previous commit
3867 We only want to count dynamic payloaders
3869 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
3871 * gst/rtsp-server/rtsp-media.c:
3872 * tests/check/gst/media.c:
3873 rtsp-media: Handle multiple dynamic elements
3874 If we have more than one dynamic payloader in the pipeline, we need
3875 to wait until the *last* one emits 'no-more-pads' before switching
3877 Failure to do so would result in a race where some of the streams
3878 wouldn't properly be prepared
3879 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3881 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
3883 * win32/common/libgstrtspserver.def:
3884 win32: Fix exported symbols list
3886 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3888 * gst/rtsp-server/rtsp-stream.c:
3889 rtsp-stream: Only update the RTP udpsink if it actually exists
3890 For send-only streams it does not exist, but the RTCP udpsink might.
3892 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
3894 * win32/common/libgstrtspserver.def:
3895 win32: Update exports
3897 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
3899 * gst/rtsp-server/rtsp-media.c:
3900 * gst/rtsp-server/rtsp-stream.c:
3901 * gst/rtsp-server/rtsp-stream.h:
3902 rtsp-media: seek on media pipelines that are complete
3903 Make sure that a seek is performed on pipelines that
3904 contain at least one sink element.
3905 Change-Id: Icf398e10add3191d104b1289de612412da326819
3906 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3908 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
3910 * gst/rtsp-server/rtsp-client.c:
3911 * gst/rtsp-server/rtsp-media.c:
3912 * gst/rtsp-server/rtsp-media.h:
3913 * gst/rtsp-server/rtsp-stream.c:
3914 * gst/rtsp-server/rtsp-stream.h:
3915 * tests/check/gst/client.c:
3916 * tests/check/gst/media.c:
3917 * tests/check/gst/rtspserver.c:
3918 * tests/check/gst/stream.c:
3919 Dynamically reconfigure pipeline in PLAY based on transports
3920 The initial pipeline does not contain specific transport
3921 elements. The receiver and the sender parts are added
3923 If the media is shared, the streams are dynamically
3924 reconfigured after each PLAY.
3925 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3927 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
3929 * gst/rtsp-server/rtsp-stream.c:
3930 rtsp-stream: obtain stream position from pad
3931 If no sinks have been added yet, obtain the current and
3932 the stop position of the stream from the send_src pad.
3933 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
3934 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3936 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
3938 * gst/rtsp-server/rtsp-session-media.c:
3939 * gst/rtsp-server/rtsp-session-media.h:
3940 rtsp-session-media: add function to get a list of transports
3941 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
3942 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3944 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
3946 * gst/rtsp-server/rtsp-stream.c:
3947 * gst/rtsp-server/rtsp-stream.h:
3948 rtsp-stream: add functions to get rtp and rtcp multicast sockets
3949 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
3950 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3952 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
3954 * gst/rtsp-server/rtsp-stream.c:
3955 stream: set async=sync=false only for RTCP appsink
3956 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
3957 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3959 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
3961 * gst/rtsp-server/rtsp-media.c:
3962 rtsp-media: return minimum value in query position case
3963 The minimum position should be returned as we are interested
3964 in the whole interval.
3965 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
3966 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3968 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
3970 * gst/rtsp-server/rtsp-session.c:
3971 * tests/check/gst/rtspserver.c:
3972 rtsp-session: Handle the case when timeout=0
3973 According to the documentation, a timeout of value 0 means
3974 that the session never timeouts. This adds handling of that.
3975 If timeout=0 we just return with a -1 from
3976 gst_rtsp_session_next_timeout_usec ().
3977 https://bugzilla.gnome.org/show_bug.cgi?id=785058
3979 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3981 * gst/rtsp-sink/gstrtspclientsink.c:
3982 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
3983 https://bugzilla.gnome.org/show_bug.cgi?id=785024
3985 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3987 * docs/libs/gst-rtsp-server-sections.txt:
3988 * gst/rtsp-server/rtsp-media-factory.c:
3989 docs: add media factory transport mode accessors
3990 and fix the documentation for the return value of the getter
3992 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
3994 * gst/rtsp-server/rtsp-client.c:
3995 rtsp-client: unref 'pipelined_requests' in finalize
3996 The hash table priv->pipelined_requests is not unref:ed in the
3997 finalize funktion. Make sure it is.
3998 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4000 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4002 * gst/rtsp-server/rtsp-media.c:
4003 rtsp-media: Initialize scalar variable
4006 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4008 * win32/common/libgstrtspserver.def:
4009 win32: Update export file
4011 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4013 * gst/rtsp-server/rtsp-client.c:
4014 * gst/rtsp-server/rtsp-media.c:
4015 * gst/rtsp-server/rtsp-media.h:
4016 Start support for RTSP 2.0
4017 This adds basic support for new 2.0 features, though the protocol is
4018 subposdely backward incompatible, most semantics are the sames.
4021 * version negotiation
4022 * pipelined requests support
4023 * Media-Properties support
4024 * Accept-Ranges support
4026 * gst_rtsp_media_seekable
4027 The RTSP methods that have been removed when using 2.0 now return
4029 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4031 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4033 * gst/rtsp-server/rtsp-stream.c:
4034 stream: Use stream duration as stream-stop if segment was not configured with a stop
4035 Allowing client to know stream duration when no seeking happened.
4036 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4038 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4040 * gst/rtsp-server/rtsp-media-factory.c:
4041 rtsp-media-factory: Don't cache any media if NULL was returned as key
4042 The docs already mentioned this, but we actually stored it in the hash
4043 table with key==NULL and leaked its reference forever.
4045 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4047 * gst/rtsp-sink/gstrtspclientsink.c:
4048 * gst/rtsp-sink/gstrtspclientsink.h:
4049 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4050 This is a simple port of:
4051 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4052 * c438545dc9e2f14f657bc0ef261fff726449867b
4053 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4054 in gst-plugins-good.
4056 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4058 * gst/rtsp-server/rtsp-sdp.c:
4059 sdp: fix Memory leak in error case
4060 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4062 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4064 * pkgconfig/meson.build:
4065 meson: don't install -uninstalled.pc file
4066 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4068 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4071 Automatic update of common submodule
4072 From 48a5d85 to 3f4aa96
4074 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4076 * gst/rtsp-server/rtsp-client.c:
4077 rtsp-client: Fix typo in debug message
4079 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4082 meson: hide symbols by default unless explicitly exported
4084 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4086 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4087 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4088 Fixes meson warning about undefined @srcdir@.
4090 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4092 * tests/meson.build:
4093 meson: skip tests on windows for now
4094 As we do in the other modules. As libgstcheck is currently not
4095 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4096 the subproject 'gstreamer' does not exist"" Meson error.
4098 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4100 * gst/rtsp-server/rtsp-stream.c:
4101 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4102 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4103 multiudpsink's last-sample always comes from the payloader. Which
4104 is wrong if auxiliary streams are multiplexed in the same stream.
4105 So check the buffer's ssrc against the caps'ssrc before to use its
4106 seqnum. If not the same ssrc just use the payloader as done prior
4107 the commit above or when there is no last-sample yet.
4108 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4110 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4113 meson: Allow using glib as a subproject
4115 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4118 meson: fix with-package-name option
4119 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4121 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4124 Distribute meson_options.txt
4126 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4129 And config.h.meson is no longer dist either
4131 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4135 meson: config.h.meson is no longer needed
4137 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4139 * tests/check/meson.build:
4140 * tests/meson.build:
4141 meson: Fix building tests and activate them again
4143 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4145 * tests/check/meson.build:
4146 meson: Do not use path separator in test names
4147 Avoiding warnings like:
4148 WARNING: Target "elements/audioamplify" has a path separator in its name.
4150 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4153 * meson_options.txt:
4154 meson: add options to set package name and origin
4155 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4157 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4159 * gst/rtsp-server/rtsp-address-pool.h:
4160 * gst/rtsp-server/rtsp-auth.h:
4161 * gst/rtsp-server/rtsp-client.h:
4162 * gst/rtsp-server/rtsp-context.h:
4163 * gst/rtsp-server/rtsp-media-factory-uri.h:
4164 * gst/rtsp-server/rtsp-media-factory.h:
4165 * gst/rtsp-server/rtsp-media.h:
4166 * gst/rtsp-server/rtsp-mount-points.h:
4167 * gst/rtsp-server/rtsp-params.h:
4168 * gst/rtsp-server/rtsp-permissions.h:
4169 * gst/rtsp-server/rtsp-sdp.h:
4170 * gst/rtsp-server/rtsp-server.h:
4171 * gst/rtsp-server/rtsp-session-media.h:
4172 * gst/rtsp-server/rtsp-session-pool.h:
4173 * gst/rtsp-server/rtsp-session.h:
4174 * gst/rtsp-server/rtsp-stream-transport.h:
4175 * gst/rtsp-server/rtsp-stream.h:
4176 * gst/rtsp-server/rtsp-thread-pool.h:
4177 * gst/rtsp-server/rtsp-token.h:
4178 Mark symbols explicitly for export with GST_EXPORT
4180 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4183 * gst/rtsp-sink/Makefile.am:
4184 Remove plugin specific static build option
4185 Static and dynamic plugins now have the same interface. The standard
4186 --enable-static/--enable-shared toggle are sufficient.
4188 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4194 === release 1.12.0 ===
4196 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4202 * gst-rtsp-server.doap:
4206 === release 1.11.91 ===
4208 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4214 * gst-rtsp-server.doap:
4218 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4221 Automatic update of common submodule
4222 From 60aeef6 to 48a5d85
4224 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4226 * gst/rtsp-server/rtsp-media-factory.c:
4227 * gst/rtsp-server/rtsp-media.c:
4228 * gst/rtsp-server/rtsp-session.c:
4229 * gst/rtsp-server/rtsp-stream.c:
4230 gi: Fix some annotations and docstrings
4232 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4234 * gst/rtsp-server/meson.build:
4236 * meson_options.txt:
4239 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4243 Automatic update of common submodule
4244 From 39ac2f5 to 60aeef6
4246 === release 1.11.90 ===
4248 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4254 * gst-rtsp-server.doap:
4258 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4260 * examples/test-launch.c:
4261 examples: make test-launch pipeline shared by default as well
4263 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4265 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4266 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4267 Just the build dir is not going to work for srcdir!=builddir.
4269 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4272 meson: Update version
4274 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4279 === release 1.11.2 ===
4281 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4287 * gst-rtsp-server.doap:
4290 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4293 meson: dist meson build files
4294 Ship meson build files in tarballs, so people who use tarballs
4295 in their builds can start playing with meson already.
4297 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4299 * examples/test-record.c:
4300 examples/test-record: Add extra line to initial printout
4301 Add an example line of how to deliver a stream to the
4304 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4306 * gst/rtsp-server/rtsp-client.c:
4307 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4308 If there is no Content-Length header, no body would be allocated and the
4309 '\0' would also not be appended to the body.
4311 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4313 * gst/rtsp-server/rtsp-client.c:
4314 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4315 While they logically have 0 bytes length, GstRTSPConnection is appending
4316 a '\0' to everything making the size be 1 instead.
4318 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4323 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4325 * gst/rtsp-server/rtsp-session.c:
4326 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4327 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4330 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4335 === release 1.11.1 ===
4337 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4343 * gst-rtsp-server.doap:
4344 * win32/common/libgstrtspserver.def:
4347 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4349 * gst/rtsp-server/rtsp-stream.c:
4350 rtsp-stream: corrected if-statement in _get_server_port()
4351 This bug was accidentally introduced while fixing a segfault
4352 in _get_server_port() function.
4353 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4355 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4357 * gst/rtsp-server/rtsp-stream.c:
4358 * tests/check/gst/stream.c:
4359 rtsp-stream: fixed segmenation fault in _get_server_port()
4360 Calling function gst_rtsp_stream_get_server_port() results in
4361 segmenation fault in the RTP/RTSP/TCP case.
4362 Port that the server will use to receive RTCP makes only
4363 sense in the UDP case, however the function should handle
4364 the TCP case in a nicer way.
4365 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4367 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4369 * gst/rtsp-server/rtsp-media-factory.c:
4370 dosc: Fix a little typo
4371 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4373 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4375 * pkgconfig/Makefile.am:
4376 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4377 * pkgconfig/meson.build:
4378 meson: generate pkg-config -uninstalled pc files
4379 Generating those files is useful for users building the GStreamer stack
4380 using meson and having to link it to another project which is still
4381 using the autotools.
4382 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4384 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4386 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4387 pkgconfig: fix -uninstalled pc file
4388 pcfiledir was never defined so the paths were wrong.
4389 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4391 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4393 * gst/rtsp-server/rtsp-stream.c:
4394 * tests/check/gst/rtspserver.c:
4395 rtsp-stream: Fixed TCP transport case
4396 Make sure that the appsink element is actually added to
4397 the bin before trying to link it with the elements in it.
4398 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4400 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4406 Remove generated .spec file
4407 Likely extremely bitrotten, and we should not ship this anyway.
4409 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4412 Automatic update of common submodule
4413 From f980fd9 to 39ac2f5
4415 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4417 * gst/rtsp-server/rtsp-media.c:
4418 media: Fix pt map caps
4419 Since decryption is handled within rtpbin, all outcoming stream
4420 caps will be application/x-rtp (i.e. regular rtp)
4421 Fixes RECORD with SRTP streams
4423 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4425 * gst/rtsp-server/rtsp-media-factory.c:
4426 media-factory: Create media objects with the proper transport mode
4427 The function called immediately afterwards (collect_streams()) will
4428 need it to work properly
4430 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4432 * gst/rtsp-server/rtsp-auth.c:
4433 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4435 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4437 * gst/rtsp-server/rtsp-media-factory.c:
4438 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4439 We're going to put a pipeline into a pipeline otherwise, which is not
4442 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4444 * gst/rtsp-server/rtsp-media.c:
4445 media: Fix race condition around finish_unprepare() if called multiple time
4446 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4448 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4450 * gst/rtsp-sink/gstrtspclientsink.c:
4451 rtspclientsink: Don't leave stale pointer after unref
4452 Fix a warning on shutdown - don't keep a pointer to an
4453 alread-unreffed object.
4455 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4458 common: use https protocol for common submodule
4459 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4461 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4463 * gst/rtsp-server/rtsp-stream.c:
4464 stream: block the output of rtpbin instead of the source pipeline
4465 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4466 detection of the srtp rollover counter to add to the SDP.
4467 Unfortunately, it was incomplete for live pipelines where the logic
4468 blocks the source bin before creating the SDP and thus would never have
4469 the necessary informaiton to create a correct SDP with srtp encryption.
4470 Move the pad blocks to rtpbin's output pads instead so that the
4471 necessary information can be created before we need the information for
4473 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4475 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4477 * gst/rtsp-server/rtsp-client.c:
4478 rtsp-client: add IDLE timeout, before session exists
4479 The RTSP server will not timeout an idle RTSP connection
4480 (note this is different from doing timeout on a RTSP
4482 At least for Apache this is a problem when running RTSP over
4483 HTTPS since it uses one of the threads (there is a rather
4484 limited number) that are available for handling requests.
4485 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4487 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4492 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4494 * gst/rtsp-server/rtsp-stream.c:
4495 rtsp-stream: Set close-socket FALSE on UDP src:es
4496 With this RTSP server can use the sockets independent on the udpsrc
4498 When the udp src is finalized it will unref socket and when g_socket
4499 is finalized the socket will be closed.
4500 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4502 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4504 * gst/rtsp-sink/gstrtspclientsink.c:
4505 rtspclientsink: Move to new helper function to parse authentication responses
4506 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4508 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4510 * examples/Makefile.am:
4511 * examples/test-auth-digest.c:
4512 * gst/rtsp-server/rtsp-auth.c:
4513 * gst/rtsp-server/rtsp-auth.h:
4514 * win32/common/libgstrtspserver.def:
4515 rtsp-auth: Add support for Digest authentication
4516 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4518 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4521 * gst/rtsp-server/meson.build:
4523 * tests/check/meson.build:
4525 * win32/common/libgstrtspserver.def:
4526 Enable building with MSVC
4527 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4529 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4532 meson: gstreamer gst_check_dep does not exist on windows
4534 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4536 * gst/rtsp-server/rtsp-client.c:
4537 client: update do_send_message to match type GstRTSPClientSendFunc
4538 This type mismatch fails building with MSVC
4539 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4541 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4543 * gst/rtsp-server/rtsp-sdp.c:
4544 rtsp-sdp: Fix indentation
4546 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4548 * gst/rtsp-server/rtsp-media.c:
4549 rtsp-media: Only signal "new-state" if the state has actually changed
4550 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4552 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4554 * gst/rtsp-server/rtsp-client.c:
4555 * gst/rtsp-server/rtsp-client.h:
4556 client: emit signal in the beginning of each rtsp request
4557 These signals let the application validate the requests, configure the
4558 media/stream in a certain way and also generate error status code in
4559 case of error or bad request.
4560 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4562 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4565 meson: update version
4567 === release 1.11.0 ===
4569 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4574 === release 1.10.0 ===
4576 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4582 * gst-rtsp-server.doap:
4585 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4587 * tests/check/gst/rtspserver.c:
4588 * tests/check/gst/stream.c:
4589 tests: try to avoid using the same ports in different tests
4590 Causes problems with client multicast tests otherwise if
4591 tests are run in parallel.
4592 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4594 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4596 * tests/check/gst/client.c:
4597 tests: client: use fail_unless_equals_foo() for better failure reporting
4599 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4601 * gst/rtsp-server/rtsp-client.c:
4602 rtsp-client: Session filter in unwatch session
4603 Call session filter with filter_session_media as paramer in
4604 client_unwatch_session if using drop_backlog = FALSE.
4605 In client_unwatch_session its allowed to grow the watchs backlog.
4606 If using drop_backlog = FALSE and the backlog is full it will cause
4607 a deadlock when setting session media state to NULL
4608 if the backlog is not allowed to grow.
4609 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4611 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4614 meson: add fallbacks for gst modules
4617 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4619 * gst/rtsp-server/rtsp-client.c:
4620 rtsp-client: Fix factory leaking in find_media() in error cases
4621 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4623 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4625 * gst/rtsp-server/rtsp-stream.c:
4626 stream: Fix randomly missing streams from SDP with dynamic elements
4627 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4628 "pad-added" signal. In that case priv->srcpad could already have its caps,
4629 and they'll be sent to priv->send_src[0] pad. That means that when it
4630 connects "notify::caps" signal, that pad could already have received its
4631 caps and the signal won't be emitted anymore.
4632 In that case priv->caps stay to NULL and when building the SDP that stream
4633 gets ignored. Leading to missing video or audio when playing in client side.
4634 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4636 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4639 meson: update version
4641 === release 1.9.90 ===
4643 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4649 * gst-rtsp-server.doap:
4652 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4654 * gst/rtsp-server/rtsp-media-factory.c:
4655 * gst/rtsp-server/rtsp-media.c:
4656 * gst/rtsp-server/rtsp-stream.c:
4657 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4658 To prevent any possibly confusion with IPs or anything else.
4659 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4661 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4663 * gst/rtsp-server/rtsp-media-factory.c:
4664 * gst/rtsp-server/rtsp-media.c:
4665 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4666 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4668 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4671 configure: Depend on gstreamer 1.9.2.1
4673 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4677 Automatic update of common submodule
4678 From b18d820 to f980fd9
4680 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4684 Automatic update of common submodule
4685 From 6f2d209 to b18d820
4687 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4689 * gst/rtsp-server/rtsp-stream.c:
4690 rtsp-stream: Remove unused _locked() variant of a function
4691 It was added during refactoring.
4693 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4695 * gst/rtsp-server/rtsp-stream.c:
4696 stream: cosmetic cleanup
4697 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4699 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4701 * gst/rtsp-server/rtsp-stream.c:
4702 stream: Compare IP addresses case insensitive in more places
4703 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4705 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4708 * gst/rtsp-server/rtsp-stream.c:
4709 stream: Fix leaked joined_bin
4710 There is no need to keep a strong ref on it, and _leave_bin() was
4711 setting it to NULL before calling g_clear_object() so it was leaked.
4712 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4714 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4716 * gst/rtsp-server/rtsp-stream.c:
4717 rtsp-stream: Compare IP address strings case insensitive
4718 Otherwise IPv6 addresses might fail this comparision.
4720 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
4722 * gst/rtsp-server/rtsp-stream.c:
4723 rtsp-stream: Bind multicast sockets to ANY as before
4724 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
4726 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
4728 * gst/rtsp-server/rtsp-session.c:
4729 rtsp-session: Fix segfault when doing keep-alive after removing the session
4730 If keep-alive happens after removing the session but before finalizing the
4731 stream transport, we would segfault.
4732 https://bugzilla.gnome.org/show_bug.cgi?id=750544
4734 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
4736 * gst/rtsp-server/rtsp-stream.c:
4737 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
4738 Adding them later will cause deadlocks due to
4739 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
4740 2) adding the multicast sink
4741 3) waiting for it to get data to preroll again
4742 3) never happens because the queues after the tee are full.
4744 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
4746 * gst/rtsp-server/rtsp-stream.c:
4747 rtsp-stream: Fix up various multicast related issues
4749 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
4751 * tests/check/gst/stream.c:
4752 tests: Fix compilation
4754 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4756 * gst/rtsp-server/rtsp-client.c:
4757 * gst/rtsp-server/rtsp-stream.c:
4758 * tests/check/gst/stream.c:
4759 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
4760 This is basically reverting changes introduced in commit f62a9a7,
4761 because it was introducing various regressions:
4762 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
4763 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
4764 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
4765 - If a mcast client connects, it creates a new socket in SETUP to try to respect
4766 the destination/port given by the client in the transport, and overrides the
4767 socket already set on the udpsink element. That means that if we already had a
4768 client connected, the source address on the udp packets it receives suddenly
4770 - If a 2nd mcast client connects, the destination/port in its transport is
4771 ignored but its transport wasn't updated.
4772 What this patch does:
4773 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
4774 - Always have a tee+queue when udp is enabled. This could be optimized
4775 again in a later patch, but is more complicated. If no unicast clients
4776 connects then those elements are useless, this could be also optimized
4778 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
4779 seperated from those for unicast clients. Since we already support only
4780 one mcast address, we also create only one set of elements.
4781 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4783 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4785 * gst/rtsp-server/rtsp-stream.c:
4786 stream: factor our plug_src function
4787 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4789 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4791 * gst/rtsp-server/rtsp-stream.c:
4792 stream: factor out plug_sink function
4793 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4795 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4797 * gst/rtsp-server/rtsp-stream.c:
4798 stream: small documentation clarification
4799 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4801 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4803 * gst/rtsp-server/rtsp-stream.c:
4804 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
4805 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4807 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4809 * gst/rtsp-server/rtsp-stream.c:
4810 stream: Keep a ref on joined bin
4811 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4813 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4815 * gst/rtsp-server/rtsp-stream.c:
4816 stream: code cleanup
4817 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4819 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4821 * gst/rtsp-server/rtsp-stream.c:
4822 stream: small fix in error code path
4823 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4825 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4827 * gst/rtsp-server/rtsp-stream.c:
4828 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
4829 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
4830 but keeps unit tests.
4831 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4833 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
4838 === release 1.9.2 ===
4840 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
4846 * gst-rtsp-server.doap:
4849 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
4852 * examples/meson.build:
4854 * gst/rtsp-server/meson.build:
4855 * gst/rtsp-sink/meson.build:
4857 * pkgconfig/meson.build:
4858 * tests/check/meson.build:
4859 * tests/meson.build:
4860 Add support for Meson as alternative/parallel build system
4861 https://github.com/mesonbuild/meson
4863 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
4866 * tests/check/Makefile.am:
4867 build: silence error about pthread for 'make check' in osx
4868 Fixes "clang: error: argument unused during compilation: '-pthread'"
4870 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
4872 * gst/rtsp-server/rtsp-client.c:
4873 rtsp-client: Fix leaking of media in error cases
4874 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
4875 and myself to make the media refcounting a bit easier to follow.
4876 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4878 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4880 * gst/rtsp-server/rtsp-client.c:
4881 rtsp-client: Fix leaking of session in error cases
4882 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4884 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
4887 Automatic update of common submodule
4888 From f363b32 to f49c55e
4890 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
4895 === release 1.9.1 ===
4897 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4903 * gst-rtsp-server.doap:
4906 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4909 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
4910 https://bugzilla.gnome.org/show_bug.cgi?id=767463
4912 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4915 Automatic update of common submodule
4916 From ac2f647 to f363b32
4918 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4920 * gst/rtsp-server/rtsp-sdp.c:
4921 * gst/rtsp-server/rtsp-sdp.h:
4922 * gst/rtsp-server/rtsp-stream.c:
4923 * gst/rtsp-server/rtsp-stream.h:
4924 sdp: add rollover counters for all sender SSRC
4925 We add different crypto sessions in MIKEY, one for each sender
4926 SSRC. Currently, all of them will have the same security policy, 0.
4927 The rollover counters are obtained from the srtpenc element using the
4929 https://bugzilla.gnome.org/show_bug.cgi?id=730539
4931 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
4933 * gst/rtsp-server/rtsp-media-factory.h:
4934 * gst/rtsp-server/rtsp-server.h:
4935 docs: fix some typos
4937 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
4939 * gst/rtsp-server/Makefile.am:
4940 g-i: pass compiler env to g-ir-scanner
4941 It's what introspection.mak does as well. Should
4942 fix spurious build failures on gnome-continuous
4943 (caused by g-ir-scanner getting compiler details
4944 via python which is broken in some environments
4945 so passing the compiler details bypasses that).
4947 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
4949 * gst/rtsp-server/rtsp-session.c:
4950 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
4951 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
4952 https://bugzilla.gnome.org/show_bug.cgi?id=766619
4954 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
4956 * gst/rtsp-sink/gstrtspclientsink.c:
4957 rtspclientsink: Check return value of sscanf
4958 And just make sure we always have 0/0 if we have an error
4961 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
4963 * gst/rtsp-server/rtsp-stream.c:
4964 * tests/check/gst/rtspserver.c:
4965 * tests/check/gst/stream.c:
4966 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
4967 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
4968 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
4969 - Create unit test for shared media.
4970 https://bugzilla.gnome.org/show_bug.cgi?id=764744
4972 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4974 * gst/rtsp-server/rtsp-stream.c:
4975 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
4976 For IPv6 addresses, binding to a multicast group does not work on Linux
4977 either. Always bind to ANY and then later join the multicast group.
4978 https://bugzilla.gnome.org/show_bug.cgi?id=764679
4980 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
4983 Automatic update of common submodule
4984 From 6f2d209 to ac2f647
4986 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
4988 * gst/rtsp-server/rtsp-thread-pool.c:
4989 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
4990 Clarified why it is necessary to add source information to
4991 GstRTSPThreadImpl. See the reported bug in GLib:
4992 https://bugzilla.gnome.org/show_bug.cgi?id=720186
4993 for more information.
4994 https://bugzilla.gnome.org/show_bug.cgi?id=761702
4996 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
4998 * examples/Makefile.am:
4999 examples: Clean up CFLAGS/LDADD even more
5000 The internal .la should come first and is part of LDADD, as is
5003 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5005 * examples/Makefile.am:
5006 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5008 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5010 * gst/rtsp-server/Makefile.am:
5011 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5013 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5015 * gst/rtsp-server/rtsp-client.c:
5016 * gst/rtsp-server/rtsp-media-factory.c:
5017 * gst/rtsp-server/rtsp-media-factory.h:
5018 * gst/rtsp-server/rtsp-media.c:
5019 * gst/rtsp-server/rtsp-media.h:
5020 * gst/rtsp-server/rtsp-sdp.c:
5021 * gst/rtsp-server/rtsp-stream.c:
5022 * gst/rtsp-server/rtsp-stream.h:
5023 rtsp-server: Implement clock signalling according to RFC7273
5024 For NTP and PTP clocks we signal the actual clock that is used and signal
5025 the direct media clock offset.
5026 For all other clocks we at least signal that it's the local sender clock.
5027 This allows receivers to know which clock was used to generate the media and
5028 its RTP timestamps. Receivers can then implement network synchronization,
5029 either absolute or at least relative by getting the sender clock rate directly
5030 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5032 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5034 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5036 * gst/rtsp-sink/gstrtspclientsink.c:
5037 rtspclientsink: Add support for setting the multicast interface
5038 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5040 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5042 * gst/rtsp-server/rtsp-media-factory.c:
5043 * gst/rtsp-server/rtsp-media-factory.h:
5044 * gst/rtsp-server/rtsp-media.c:
5045 * gst/rtsp-server/rtsp-media.h:
5046 * gst/rtsp-server/rtsp-stream.c:
5047 * gst/rtsp-server/rtsp-stream.h:
5048 rtsp-media: Add support for setting the multicast interface
5049 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5051 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5053 * gst/rtsp-sink/gstrtspclientsink.c:
5054 rtspclientsink: use new gst_element_class_add_static_pad_template()
5055 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5057 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5062 === release 1.8.0 ===
5064 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5070 * gst-rtsp-server.doap:
5073 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5075 * gst/rtsp-server/rtsp-stream.c:
5076 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5077 This would get us NO_PREROLL in the bin again and break seeking.
5078 Thanks to Carlos Rafael Giani for helping to debug this!
5079 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5081 === release 1.7.91 ===
5083 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5089 * gst-rtsp-server.doap:
5092 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5094 * gst/rtsp-server/rtsp-stream.c:
5095 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5096 Without this, RECORD pipelines are broken because
5097 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5098 added later. Previously it was there earlier and due to NO_PREROLL caused the
5099 pipeline to preroll immediately
5100 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5101 as the corresponding code previously was only for PLAY pipelines.
5102 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5104 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5106 * gst/rtsp-server/rtsp-stream.c:
5107 rtsp-stream: Fix typo in the docstring
5108 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5110 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5112 * gst/rtsp-server/rtsp-stream.c:
5113 rtsp-stream: Disable multicast loopback for all our sockets
5114 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5115 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5116 loopback setting on the socket... while udpsink does which unfortunately has
5117 no effect here on Windows but on Linux.
5118 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5120 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5122 * tests/check/gst/stream.c:
5123 stream tests: added new tests
5124 Test a case when the address pool only contains multicast addresses
5125 and the client is requesting unicast udp.
5126 Added tests for multicast ports allocation.
5127 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5129 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5131 * gst/rtsp-server/rtsp-stream.c:
5132 rtsp-stream: Only bind multicast sockets to ANY on Windows
5133 On Linux it is still needed to bind to the multicast address
5134 to filter out random other packets, while on Windows binding
5135 to multicast addresses just fails.
5137 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5139 * gst/rtsp-server/rtsp-stream.c:
5140 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5141 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5142 addresses, which is something that used to work before. For unicast addresses
5143 if the pool contains none, we just allocate them as if there is no pool at
5145 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5147 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5149 * gst/rtsp-server/rtsp-client.c:
5150 * gst/rtsp-server/rtsp-stream.c:
5151 rtsp-server: Fix indentation
5153 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5155 * gst/rtsp-server/rtsp-stream.c:
5156 rtsp-stream: Don't bind the sockets to multicast addresses
5157 This works on Linux but fails completely on Windows. You're supposed
5158 to bind to ANY and then join the multicast group.
5159 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5161 === release 1.7.90 ===
5163 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5169 * gst-rtsp-server.doap:
5172 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5175 Automatic update of common submodule
5176 From b64f03f to 6f2d209
5178 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5180 * gst/rtsp-sink/gstrtspclientsink.c:
5181 * tests/check/gst/rtspclientsink.c:
5182 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5183 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5185 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5187 * tests/check/gst/media.c:
5188 * tests/check/gst/rtspclientsink.c:
5189 * tests/check/gst/rtspserver.c:
5190 * tests/check/gst/stream.c:
5191 tests: unit test fixes
5192 Removed port allocation test from the media suite.
5193 The port allocation failure is now in the stream suite.
5195 Make sure that the media is suspended after the DESCRIBE request
5196 before reconfiguring the UDP sinks.
5198 In the RECORD case we have to set async property to false
5199 for the appsink element in the test in order to make sure
5200 that the media pipeline doesn't hang in start_preroll().
5201 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5203 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5205 * gst/rtsp-server/rtsp-client.c:
5206 * gst/rtsp-server/rtsp-stream.c:
5207 * gst/rtsp-server/rtsp-stream.h:
5208 rtsp-stream: postpone UDP socket allocation until SETUP
5209 Postpone the allocation of the UDP sockets until we know
5210 what transport has been chosen by the client.
5211 Both unicast and multicast UDP sources are created in one
5213 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5215 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5217 * gst/rtsp-server/rtsp-stream.c:
5218 rtsp-stream: postpone the creation of the UDP sources
5219 Code refactoring: allocate the UDP ports after the sender and
5220 the reciver parts have been created.
5221 We postpone the creation of the UDP sources until the UDP
5222 ports have been allocated.
5223 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5225 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5227 * gst/rtsp-server/rtsp-stream.c:
5228 rtsp-stream: added function for setting UDP sources to PLAYING state
5229 Code refactoring: Introduced a function for setting UDP sources
5231 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5233 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5235 * gst/rtsp-server/rtsp-stream.c:
5236 rtsp-stream: added function for creating and configuring UDP sources
5237 Code refactoring: create and configure UDP sources in a separate function.
5238 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5240 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5242 * gst/rtsp-server/rtsp-stream.c:
5243 rtsp-stream: added function for RTP/RTCP socket configuration
5244 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5245 in a separate function.
5246 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5248 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5250 * gst/rtsp-server/rtsp-stream.c:
5251 rtsp-stream: added function for creating and configuring UDP sinks
5252 Code refactoring: create and configure UDP sinks in a separate function.
5253 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5255 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5257 * gst/rtsp-server/rtsp-stream.c:
5258 rtsp-stream: added helper function for creating the sender/receiver parts
5259 Code refactoring: introduced helper function for creating
5260 the receiver and the sender parts of the streaming pipeline.
5261 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5263 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5268 === release 1.7.2 ===
5270 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5276 * gst-rtsp-server.doap:
5279 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5281 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5282 uninstalled.pc: add support for non libtool build systems
5283 Currently the .la path is provided which requires to use libtool as
5284 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5285 It is fine as long as the application is built using libtool.
5286 So currently it is not possible to compile a GStreamer application
5287 within gst-uninstalled with CMake or other build system different
5289 This patch allows to do the following in gst-uninstalled env:
5290 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5291 gstreamer-rtsp-server-1.0)
5292 Previously it required to prepend libtool --mode=link
5293 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5295 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5297 * gst/rtsp-sink/gstrtspclientsink.c:
5298 rtspclientsink: remove check for impossible condition
5299 Goto error label checks stream to see if it needs to be unreferenced before
5300 returning, but this goto jumps happens before the stream is ever set, so it
5301 will always be NULL in this error label.
5304 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5306 * gst/rtsp-sink/gstrtspclientsink.c:
5307 rtspclientsink: clean switch statements
5308 Coverity demands for fallthrough statements to be clearly commented,
5309 to distinguish from accidental fall throughs. And it also needs all
5310 cases to finish with a break, even if the break is never going to be
5311 executed like in the case of a continue jump.
5315 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5317 * tests/check/Makefile.am:
5318 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5319 To get the CK_DEFAULT_TIMEOUT defined for all tests
5320 Also removes a 120 seconds timeout that was set as default
5321 explicitly in this module
5322 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5324 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5328 Automatic update of common submodule
5329 From 86e4663 to b64f03f
5331 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5333 * gst/rtsp-server/rtsp-media.c:
5334 rtsp-media: fix state_lock not locked again when preroll fails
5335 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5337 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5340 configure: Move plugin specific flags below all the others
5341 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5342 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5344 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5346 * gst/rtsp-sink/gstrtspclientsink.c:
5347 rtspclientsink: Simplify slightly using new -base API
5348 Use the new Mikey and SDP API in the base plugins libs
5349 to simplify some code.
5350 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5352 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5357 * gst/rtsp-sink/Makefile.am:
5358 * gst/rtsp-sink/gstrtspclientsink.c:
5359 * gst/rtsp-sink/gstrtspclientsink.h:
5360 * gst/rtsp-sink/plugin.c:
5361 * tests/check/Makefile.am:
5362 * tests/check/gst/rtspclientsink.c:
5363 rtspsink: Add rtspclientsink element
5364 Add an rtspclientsink element that accepts streams for which
5365 there is a registered payloader and sends them to
5366 an RTSP server using RECORD.
5367 Sending is synchronised to the pipeline clock. Payload-types
5368 are automatically selected. The 'new-payloader' signal is fired
5369 for custom configuration of payloaders when they are created.
5370 Can now stream a movie like this:
5372 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5373 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5375 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5376 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5377 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5379 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5381 * gst/rtsp-server/rtsp-stream.c:
5382 * gst/rtsp-server/rtsp-stream.h:
5383 rtsp-stream: Add functions for using rtsp-stream from the client
5384 Add a boolean to indicate that the rtsp-stream is running on the
5385 'client' side of an RTSP connection, for sending streams via
5386 RECORD. In that case, the roles of the client/server ports
5387 in transport setup are swapped.
5388 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5390 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5392 * gst/rtsp-server/rtsp-sdp.c:
5393 * gst/rtsp-server/rtsp-sdp.h:
5394 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5395 A new function that adds info from a GstRTSPStream into an SDP message.
5396 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5398 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5400 * gst/rtsp-server/rtsp-media.c:
5401 rtsp-media: Fix mutex beeing unlocked while they should be locked
5402 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5404 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5406 * gst/rtsp-server/rtsp-media-factory.c:
5407 rtsp-media-factory: add missing break in "clock" property setter
5410 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5412 * gst/rtsp-server/rtsp-stream.c:
5413 rtsp-stream: fixed assert during update transport
5414 When RTSP server trying update transport during multicast, it throws an
5415 assert. The assert is thrown because it is trying to get the parent of
5416 an non-existing funnel element.
5417 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5419 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5421 * gst/rtsp-server/rtsp-permissions.h:
5422 * gst/rtsp-server/rtsp-thread-pool.h:
5423 * gst/rtsp-server/rtsp-token.h:
5424 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5425 gtk-doc can handle static inline functions just fine these days,
5426 there's no need for this stuff any more.
5428 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5430 * gst/rtsp-server/rtsp-media.c:
5431 * gst/rtsp-server/rtsp-sdp.c:
5432 sdp: replace duplicated codes to call new base sdp apis
5433 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5435 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5437 * examples/test-netclock.c:
5438 test-netclock: Use the new API to configure a clock directly
5440 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5442 * gst/rtsp-server/rtsp-media-factory.c:
5443 * gst/rtsp-server/rtsp-media-factory.h:
5444 * gst/rtsp-server/rtsp-media.c:
5445 * gst/rtsp-server/rtsp-media.h:
5446 rtsp-media: Add API to directly configure a clock on the media pipelines
5448 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5450 * gst/rtsp-server/rtsp-media.c:
5451 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5453 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5455 * gst/rtsp-server/rtsp-media-factory.c:
5456 rtsp-media-factory: Add FIXME for 2.0
5458 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5460 * gst/rtsp-server/rtsp-stream.c:
5461 rtsp-stream: Fix indentation
5463 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5465 * gst/rtsp-server/rtsp-media.c:
5466 rtsp-media: Do not prepare media after media times out
5467 Deferred calls to start_prepare() can be deferred past the point until
5468 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5469 prepared to wait. Previously there was no lock and no check for this
5470 situation. This meant that a media could be prepared and unprepared
5471 simultaneously by two different threads. Now a lock is in place and a
5472 suitable check is done.
5473 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5475 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5477 * gst/rtsp-server/rtsp-client.c:
5478 * gst/rtsp-server/rtsp-media-factory.c:
5479 * gst/rtsp-server/rtsp-media-factory.h:
5480 * gst/rtsp-server/rtsp-media.c:
5481 * gst/rtsp-server/rtsp-media.h:
5482 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5483 Without TEARDOWN it might be desireable to keep the media running and continue
5484 sending data to the client, even if the RTSP connection itself is
5486 Only do this for session medias that have only UDP transports. If there's at
5487 least on TCP transport, it will stop working and cause problems when the
5488 connection is disconnected.
5489 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5491 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5496 === release 1.7.1 ===
5498 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5504 * gst-rtsp-server.doap:
5507 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5510 configure: Make -Bsymbolic check work with clang.
5511 Update the -Bsymbolic check with the version glib has. This version
5513 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5515 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5517 * gst/rtsp-server/rtsp-session-pool.c:
5518 rtsp-session-pool: Avoid dollar sign ($) in session ids
5519 Live555 in VLC strips off dollar signs and then gets very confused,
5520 we don't loose too much entropy by just skipping it.
5522 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5524 * gst/rtsp-server/rtsp-address-pool.h:
5525 * gst/rtsp-server/rtsp-auth.h:
5526 * gst/rtsp-server/rtsp-client.h:
5527 * gst/rtsp-server/rtsp-media-factory-uri.h:
5528 * gst/rtsp-server/rtsp-media-factory.h:
5529 * gst/rtsp-server/rtsp-media.h:
5530 * gst/rtsp-server/rtsp-mount-points.h:
5531 * gst/rtsp-server/rtsp-permissions.h:
5532 * gst/rtsp-server/rtsp-server.h:
5533 * gst/rtsp-server/rtsp-session-media.h:
5534 * gst/rtsp-server/rtsp-session-pool.h:
5535 * gst/rtsp-server/rtsp-session.h:
5536 * gst/rtsp-server/rtsp-stream-transport.h:
5537 * gst/rtsp-server/rtsp-stream.h:
5538 * gst/rtsp-server/rtsp-thread-pool.h:
5539 * gst/rtsp-server/rtsp-token.h:
5540 rtsp-server: Add g_autoptr() support to all types
5541 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5543 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5545 * gst/rtsp-server/rtsp-stream.c:
5546 rtsp-stream: fixed valgrind error
5547 Fixed the valgrind error in unit test. The UDP source created during
5548 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5550 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5552 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5556 Automatic update of common submodule
5557 From b319909 to 86e4663
5559 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5561 * gst/rtsp-server/rtsp-client.c:
5562 rtsp-client: suspend media during setup request
5563 SETUP request from clients needs to suspend the media to clear the
5564 prerolled buffers. Otherwise it will not affect the prerolled buffer
5565 and the prerolled buffers will be incorrect (for example block-size
5566 from setup request will not affect the prerolled buffer unless the
5567 media is suspended).
5568 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5570 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5572 * gst/rtsp-server/rtsp-stream.c:
5573 rtsp-stream: create stream pipeline based on transport
5574 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5575 only UDP is set as the transport protocol, it will not add the extra tee
5576 or queue element to the pipeline. Both these elements will be added, if
5577 it supports both TCP and UDP protocols. This improves the pipeline
5578 performance when one protocol is present.
5579 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5581 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5583 * gst/rtsp-server/rtsp-stream.c:
5584 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5585 Adding them when not needed will start some logic inside rtpbin that might be
5586 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5587 would start up a rtpjitterbuffer and behave in weird ways.
5588 We still set up the UDP sources for RTP receiving for a sender media to be
5589 able to receive any packets sent by the client for NAT traversal. They will
5590 all go to a fakesink though.
5591 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5592 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5593 receive ASYNC_DONE after a seek.
5594 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5596 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5598 * gst/rtsp-server/rtsp-stream.c:
5599 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5600 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5601 Previously we were only setting this for sender sockets, which caused looped
5602 back packets to be received on Windows if a multicast transport was used.
5604 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5606 * examples/test-record-auth.c:
5607 * examples/test-record.c:
5608 examples: Actually use the provided port in the record examples
5610 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5612 * examples/test-record-auth.c:
5613 test-record-auth: Add the option to build in TLS support
5615 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5617 * examples/test-auth.c:
5618 test-auth: Use an 'anonymous' user for unauthenticated default
5619 There's a comment on one of the resources that 'user' and 'admin'
5620 shouldn't even be able to see it, but they can if the default
5621 token is 'admin2', since that gives them access anyway.
5623 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5625 * examples/.gitignore:
5626 * examples/Makefile.am:
5627 * examples/test-record-auth.c:
5628 Add test-record-auth example
5630 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5632 * gst/rtsp-server/rtsp-client.c:
5633 * tests/check/gst/client.c:
5634 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5636 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5638 * gst/rtsp-server/rtsp-server.c:
5639 rtsp-server: Change the logic so we don't pop a NULL context
5640 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5641 will sometimes fail. This call is made before any context is pushed
5642 resulting in an attempt to pop a NULL context.
5643 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5645 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5647 * tests/check/gst/rtspserver.c:
5648 rtspserver: Add udp-mcast transport SETUP test
5649 Refactor utility functions in the test file so they can handle
5650 more than UDP and TCP as lower transport.
5651 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5653 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5655 * gst/rtsp-server/rtsp-stream.c:
5656 rtsp-stream: Always unref return value of gst_object_get_parent()
5657 Fixes a leak of a GstBin in the udp-mcast case.
5658 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5660 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5663 Automatic update of common submodule
5664 From b99800a to b319909
5666 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5669 Use new GST_ENABLE_EXTRA_CHECKS #define
5670 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5672 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5675 Automatic update of common submodule
5676 From 6babecd to b99800a
5678 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5681 Update GLib dependency to 2.40.0
5683 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5685 * examples/test-mp4.c:
5686 * gst/rtsp-server/rtsp-stream.c:
5687 stream: listen to sender ssrc signals
5688 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5690 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5693 common: update for new suppression
5694 Makes check-valgrind pass with glib 2.46
5696 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5698 * gst/rtsp-server/rtsp-media.c:
5699 rtsp-media: Take reference to media that will be prepared
5700 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5701 Later on a g_idle_source_new() is created and a pointer to the media
5702 object is passed as user data. If the media is freed before the idle
5703 source is dispatched the media object pointer is invalid, but the idle
5704 source callback expects it to still be valid. To fix this a reference to
5705 the media object is taken when registering the source callback function
5706 and a corresponding release of the reference is done when the souce is
5708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5710 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5712 * examples/test-launch.c:
5713 * examples/test-mp4.c:
5714 * examples/test-ogg.c:
5715 * examples/test-record.c:
5716 * examples/test-uri.c:
5717 rtsp-server: Fix memory leaks when context parse fails
5718 When g_option_context_parse fails, context and error variables are not getting free'd
5719 which results in memory leaks. Free'ing the same.
5720 And replacing g_error_free with g_clear_error, which checks if the error being passed
5721 is not NULL and sets the variable to NULL on free'ing.
5722 https://bugzilla.gnome.org/show_bug.cgi?id=753863
5724 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5729 === release 1.6.0 ===
5731 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5737 * gst-rtsp-server.doap:
5740 === release 1.5.91 ===
5742 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
5748 * gst-rtsp-server.doap:
5751 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
5753 * docs/libs/gst-rtsp-server-sections.txt:
5754 * gst/rtsp-server/rtsp-stream.c:
5755 stream: fix docs for recently-added get/set_buffer_size API
5756 https://bugzilla.gnome.org/show_bug.cgi?id=749095
5758 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
5760 * gst/rtsp-server/rtsp-media.c:
5761 rtsp-media: Don't crash on encrypted RTX SDP
5762 In parse_keymgmt(), don't mutate the input string that's been passed
5763 as const, especially since we might need the original value again if
5764 the same key info applies to multiple streams (RTX, for example).
5765 https://bugzilla.gnome.org/show_bug.cgi?id=754753
5767 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
5769 * examples/test-mp4.c:
5770 test-mp4: Support filenames with spaces in them. Error out on too few arguments
5772 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
5774 * examples/test-record.c:
5775 test-record: Check parameter count and print out help
5776 If no launch pipeline was supplied, print out some help
5778 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
5780 * gst/rtsp-server/rtsp-media.c:
5781 * gst/rtsp-server/rtsp-stream.c:
5782 * gst/rtsp-server/rtsp-stream.h:
5783 rtsp-stream: Implement UDP buffer size setting.
5784 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
5786 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
5787 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
5789 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
5791 * gst/rtsp-server/rtsp-media.h:
5792 rtsp-media: Fix small typo causing gtk-doc to complain
5794 === release 1.5.90 ===
5796 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5802 * gst-rtsp-server.doap:
5805 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5807 * gst/rtsp-server/rtsp-media-factory.c:
5808 media-factory: get port number through gst_rtsp_url_get_port
5809 https://bugzilla.gnome.org/show_bug.cgi?id=753473
5811 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
5813 * tests/check/gst/media.c:
5814 media-test: Removing unnecessary assertion
5815 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5817 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5819 * gst/rtsp-server/rtsp-server.c:
5820 Document that source keeps a ref on server until it's destroyed
5821 https://bugzilla.gnome.org/show_bug.cgi?id=749227
5823 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5825 * tests/check/gst/media.c:
5826 media-test: Test for multiple dynamic payload
5827 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5829 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5831 * gst/rtsp-server/rtsp-media.c:
5832 media: Only add fakesink once per pipeline
5833 The intention is to prevent going PLAYING state before pads are created.
5834 If there was mutilple dynamic payload, it would leak few fakesink and
5835 actually prevent from ever reaching playing state.
5836 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5838 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5840 * gst/rtsp-server/rtsp-media.c:
5841 Revert "rtsp-media: Only add 1 fakesink per pipeline"
5842 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
5844 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5846 * gst/rtsp-server/rtsp-media.c:
5847 rtsp-media: Only add 1 fakesink per pipeline
5848 There should be only one fakesink per pipeline, not per dynpay. This
5849 would lead to element naming clash.
5851 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
5853 * gst/rtsp-server/rtsp-media.c:
5854 rtsp-media: assertion error due to wrong condition check
5855 In media to caps function, reserved_keys array is being used for variable i,
5856 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
5857 changed it to variable j
5858 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5860 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
5862 * gst/rtsp-server/rtsp-media.c:
5863 rtsp-media: Strip keys from the fmtp that we use internally in our caps
5864 Skip keys from the fmtp, which we already use ourselves for the
5865 caps. Some software is adding random things like clock-rate into
5866 the fmtp, and we would otherwise here set a string-typed clock-rate
5867 in the caps... and thus fail to create valid RTP caps
5868 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5870 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5872 * gst/rtsp-server/rtsp-thread-pool.c:
5873 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
5874 https://bugzilla.gnome.org/show_bug.cgi?id=752640
5876 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
5879 Automatic update of common submodule
5880 From f74b2df to 9aed1d7
5882 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
5887 === release 1.5.2 ===
5889 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
5895 * gst-rtsp-server.doap:
5898 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
5900 * gst/rtsp-server/rtsp-client.c:
5901 * gst/rtsp-server/rtsp-client.h:
5902 * tests/check/gst/client.c:
5903 rtsp-client: allow application to decide what requirements are supported
5904 Add "check-requirements" signal and vfunc to allow application
5905 (and subclasses) to check the requirements.
5906 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
5907 https://bugzilla.gnome.org/show_bug.cgi?id=749417
5909 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5912 Automatic update of common submodule
5913 From 6015d26 to f74b2df
5915 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5917 * gst/rtsp-server/rtsp-media.c:
5918 rtsp-media: Always use real payloader when creating streams
5919 A bin that contains the real payloader might be used as payloader. In this
5920 case we have to get the real payloader for the various properties it provides.
5921 Example use cases for this are bins that payload some media and then have
5922 additional elements that add metadata or RTP extension headers to the stream.
5923 https://bugzilla.gnome.org/show_bug.cgi?id=750800
5925 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5927 * examples/test-netclock-client.c:
5928 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
5930 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
5932 * examples/test-netclock-client.c:
5933 * examples/test-netclock.c:
5934 test-netclock: Use new ntp-time-source property on rtpbin
5935 Select the clock time to be used as NTP time source. This allows proper
5936 synchronization between receivers, independent of sharing base times, and just
5937 requires them to use the same clock.
5939 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5941 * examples/test-netclock-client.c:
5942 * examples/test-netclock.c:
5943 test-netclock: Setting the same base time on sender and receiver is not necessary
5944 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
5946 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5948 * gst/rtsp-server/rtsp-stream.c:
5949 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
5950 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5952 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5954 * docs/libs/gst-rtsp-server.types:
5955 docs: add missing types
5956 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5958 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5960 * docs/libs/gst-rtsp-server-sections.txt:
5961 docs: add missing apis
5962 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5964 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5966 * examples/test-netclock-client.c:
5967 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
5969 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5971 * docs/libs/gst-rtsp-server-sections.txt:
5972 * gst/rtsp-server/rtsp-auth.c:
5973 * gst/rtsp-server/rtsp-auth.h:
5974 GstRTSPAuth: Add client certificate authentication support
5975 https://bugzilla.gnome.org/show_bug.cgi?id=750471
5977 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5979 * examples/test-netclock-client.c:
5980 test-netclock-client: Use new GstClock API to wait for clock synchronization
5982 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
5984 * examples/test-netclock-client.c:
5985 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
5986 A mainloop is needed to get glimagesink to display something on OSX, and
5987 the source-setup signal just makes things a little bit easier.
5989 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
5992 Automatic update of common submodule
5993 From d9a3353 to 6015d26
5995 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
5998 Automatic update of common submodule
5999 From d37af32 to d9a3353
6001 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6004 Automatic update of common submodule
6005 From 21ba2e5 to d37af32
6007 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6010 Automatic update of common submodule
6011 From c408583 to 21ba2e5
6013 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6015 * docs/libs/Makefile.am:
6016 docs: remove variables that we define in the snippet from common
6017 This is syncing our Makefile.am with upstream gtkdoc.
6019 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6022 Automatic update of common submodule
6023 From 44a3517 to c408583
6025 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6030 === release 1.5.1 ===
6032 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6038 * gst-rtsp-server.doap:
6041 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6043 * gst/rtsp-server/rtsp-client.c:
6044 rtsp-client: No flush during Teardown.
6045 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6046 backlog is empty it can happen that just a part of a message will be
6047 sent and rest is in backlog queue. If then flush during teardown
6048 just a part of message will be sent.This can lead to client miss
6049 teardown response since it expect to get the last part of message.
6050 The flushing during teardown was introduced to fix a deadlock that now
6051 is fixed more generally in handle_request by temporary setting backlog
6053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6055 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6057 * tests/check/Makefile.am:
6058 tests: Use AM_TESTS_ENVIRONMENT
6059 Needed by the new automake test runner and the
6060 current version of the common submodule.
6062 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6064 * gst/rtsp-server/rtsp-media.h:
6065 * gst/rtsp-server/rtsp-stream.h:
6066 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6068 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6070 * gst/rtsp-server/rtsp-media.c:
6071 rtsp-media: Mark some more functions static
6073 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6075 * gst/rtsp-server/rtsp-media.c:
6076 rtsp-media: Only unblock the media in suspend() when actually changing the state
6077 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6079 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6081 * examples/test-video-rtx.c:
6082 examples: Use AVPF profile for the RTX example
6084 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6086 * gst/rtsp-server/rtsp-sdp.c:
6087 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6089 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6091 * gst/rtsp-server/rtsp-stream.c:
6092 rtsp-stream: get valid clock-rate from last-sample
6093 clock-rate in last-sample's caps is integer, not unsigned.
6094 To get this value properly, variable needs to be type-casted to int.
6095 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6097 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6101 autogen.sh: only run autopoint if gettext requested in configure.ac
6102 Not just because there happens to be a po directory.
6103 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6105 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6108 Revert "configure.ac: uncomment gettext version setup"
6109 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6110 We don't need a gettext setup here and there's no po
6111 directory either, so no reason why autopoint would be
6112 run in the first place.
6113 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6115 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6117 * examples/test-multicast.c:
6118 * examples/test-multicast2.c:
6119 * examples/test-sdp.c:
6120 * examples/test-video-rtx.c:
6121 * examples/test-video.c:
6122 * tests/test-cleanup.c:
6123 * tests/test-reuse.c:
6124 Fix timeout function signatures across tests and examples
6126 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6128 * tests/check/Makefile.am:
6129 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6130 Make sure the test environment is set up.
6131 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6133 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6136 configure: bump automake requirement to 1.14 and autoconf to 2.69
6137 This is only required for builds from git, people can still
6138 build tarballs if they only have older autotools.
6139 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6141 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6144 configure.ac: uncomment gettext version setup
6145 Fixes autogen.sh. It would run autopoint, which would complain
6146 that it could not find the gettext version in configure.ac.
6147 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6149 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6151 * examples/test-video-rtx.c:
6152 test-video-rtx: set exact payload type to PCMA payloader
6153 Setting wrong payload type causes failure to do retransmission through audio stream
6154 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6156 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6158 * gst/rtsp-server/rtsp-media.c:
6159 * gst/rtsp-server/rtsp-stream.c:
6160 * gst/rtsp-server/rtsp-stream.h:
6161 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6162 Because of duplicated g_signal_connect for request-aux-sender signal,
6163 wrong stream pointer is passed to the signal handler.
6164 Instead of passing each stream, pass stream array and get the relevant stream.
6165 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6167 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6171 Update autogen.sh to latest version from common
6172 Fixes build after aclocal_check etc. helpers have been removed.
6174 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6177 Automatic update of common submodule
6178 From bc76a8b to c8fb372
6180 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6182 * gst/rtsp-server/rtsp-stream.c:
6183 rtsp-stream: Limit the queues to 1 buffer
6184 We only need them to be able to pre-roll, queueing up more data here
6185 is only going to harm latency and memory usage.
6187 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6189 * gst/rtsp-server/rtsp-stream.c:
6190 rtsp-stream: Update comment and ASCII art to the latest code
6191 We have a queue in front of the udpsink too to prevent the pipeline from
6194 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6196 * gst/rtsp-server/rtsp-stream.c:
6197 rtsp-media: Properly return first rtptime
6198 Instead we where returning first GstBuffer timestamp. This would result
6199 in clock skew and unwanted behaviour in RTSP playback.
6200 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6202 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6204 * gst/rtsp-server/rtsp-stream.c:
6205 rtsp-stream: Don't leave buffer mapped
6206 If the seq is NULL, the RTP buffer was left mapped. We should always
6209 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6214 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6216 * gst/rtsp-server/rtsp-media-factory.c:
6217 * tests/check/gst/client.c:
6218 Fix double semicolons
6220 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6222 * gst/rtsp-server/rtsp-stream.c:
6223 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6224 This gives more accurate values than asking the payloader. There might be
6225 queueing happening between the payloader and the sink.
6226 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6228 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6230 * gst/rtsp-server/rtsp-media.c:
6231 rtsp-media: Don't seek for PLAY if the position will not change
6232 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6234 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6236 * gst/rtsp-server/rtsp-media.c:
6237 rtsp-media: Don't include payload type in the caps for framesize
6238 When the sdp media attribute framesize are converted to caps
6239 the <payload> should not be included.
6240 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6241 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6243 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6245 * gst/rtsp-server/rtsp-sdp.c:
6246 rtsp-sdp: add payload type to the sdp framesize attribute
6247 The sdp framesize attribute is desribed in RFC6064. It is specified
6248 for payloading of H263 and has the following form
6249 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6250 should be added to the caps in a payloader and the <payload type> should
6251 be added by the rtsp-server.
6252 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6254 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6256 * examples/test-uri.c:
6257 examples: test-uri: fix tainted variable
6258 Insignificant but this keeps Coverity happy.
6261 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6263 * examples/.gitignore:
6264 * examples/Makefile.am:
6265 * examples/test-netclock-client.c:
6266 * examples/test-netclock.c:
6267 examples: Add a simple example of network synch for live streams.
6268 An example server and client that works for synchronising live streams
6269 only - as it can't support pause/play.
6271 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6273 * gst/rtsp-server/rtsp-media-factory.c:
6274 * gst/rtsp-server/rtsp-media-factory.h:
6275 rtsp-media-factory: Add functions to set/get the media gtype
6276 Allow specifying the GType of a GstRtspMedia subclass to create
6277 as a simpler way to get the factory to create a custom
6278 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6280 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6282 * gst/rtsp-server/rtsp-media.c:
6283 rtsp-media: fix double unlock in _get_buffer_size()
6284 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6285 because of double g_mutex_unlock () usage.
6286 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6288 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6290 * gst/rtsp-server/rtsp-session-pool.c:
6291 * gst/rtsp-server/rtsp-session.c:
6292 * gst/rtsp-server/rtsp-session.h:
6293 rtsp-session: Use monotonic time for RTSP session timeout
6294 Changed RTSP session timeout handling to monotonic time
6295 and deprecating the API for current system time.
6296 This fixes timeouts when the system time changes.
6297 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6299 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6301 * gst/rtsp-server/rtsp-client.c:
6302 * gst/rtsp-server/rtsp-media.c:
6303 rtsp-client: Only error out in PLAY if seeking actually failed
6304 If the media was just not seekable, we continue from whatever position we are
6305 and let the client decide if that is what is wanted or not.
6306 Only if the actual seek failed, we can't really recover and should error out.
6308 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6310 * gst/rtsp-server/rtsp-stream.c:
6311 rtsp-stream: Add necessary queues between tee and multiudpsink
6312 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6314 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6316 * gst/rtsp-server/rtsp-client.c:
6317 * gst/rtsp-server/rtsp-media.c:
6318 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6319 Instead error out properly the same way as if the SEEKING query already
6322 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6324 * gst/rtsp-server/rtsp-stream.h:
6325 rtsp-stream: minor code formatting fix
6327 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6329 * gst/rtsp-server/rtsp-media.c:
6330 rtsp-media: fix logic for collect_streams
6331 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6332 all streams it knows if it got any, and can check if the transport mode is OK.
6335 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6337 * gst/rtsp-server/rtsp-media.c:
6338 rtsp-media: Don't set the transport mode based on what elements we find
6339 Just print a warning if the one that was set before disagrees with what
6340 elements we found. It must already be set to something before as this
6341 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6342 and we would reject ANNOUNCE if the RECORD flag was not set.
6344 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6346 * tests/check/gst/rtspserver.c:
6347 tests: rtspserver: rename shadowed variable
6348 We have two different 'sink' variables here,
6349 rename one of them for clarity.
6351 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6353 * gst/rtsp-server/rtsp-client.c:
6354 rtsp-client: fix awkward if clause
6356 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6358 * examples/test-uri.c:
6359 examples: test-uri: improve uri argument handling and accept file names
6360 Print an error if the argument passed is not a URI and can't
6361 be converted into one, or no arguments have been provided.
6363 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6365 * examples/test-uri.c:
6366 examples: test-uri: don't remove mount point after 10 seconds
6367 It's very irritating when trying to test stuff repeatedly
6368 and serves no real purpose other than showing that it can
6371 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6373 * examples/.gitignore:
6374 examples: add new test-record to .gitignore
6376 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6378 * examples/test-record.c:
6379 * gst/rtsp-server/rtsp-client.c:
6380 * gst/rtsp-server/rtsp-media-factory.c:
6381 * gst/rtsp-server/rtsp-media-factory.h:
6382 * gst/rtsp-server/rtsp-media.c:
6383 * gst/rtsp-server/rtsp-media.h:
6384 * tests/check/gst/rtspserver.c:
6385 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6387 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6389 * examples/test-record.c:
6390 test-record: Set latency for playback-style example to 2s instead of 200ms
6392 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6394 * tests/check/gst/rtspserver.c:
6395 tests: add some unit tests for ANNOUNCE and RECORD
6396 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6398 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6400 * gst/rtsp-server/rtsp-client.c:
6401 rtsp-client: fix a couple of leaks in handle_announce
6403 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6405 * gst/rtsp-server/rtsp-media-factory.c:
6406 * gst/rtsp-server/rtsp-media-factory.h:
6407 * gst/rtsp-server/rtsp-media.c:
6408 * gst/rtsp-server/rtsp-media.h:
6409 rtsp-media: Expose latency setting for setting the rtpbin latency
6411 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6413 * examples/test-record.c:
6414 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6416 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6418 * gst/rtsp-server/rtsp-stream.c:
6419 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6421 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6423 * examples/Makefile.am:
6424 * examples/test-record.c:
6425 * gst/rtsp-server/rtsp-client.c:
6426 * gst/rtsp-server/rtsp-client.h:
6427 * gst/rtsp-server/rtsp-media-factory.c:
6428 * gst/rtsp-server/rtsp-media-factory.h:
6429 * gst/rtsp-server/rtsp-media.c:
6430 * gst/rtsp-server/rtsp-media.h:
6431 * gst/rtsp-server/rtsp-session-media.c:
6432 * gst/rtsp-server/rtsp-stream.c:
6433 * gst/rtsp-server/rtsp-stream.h:
6434 Add initial support for RECORD
6435 We currently only support media that is RECORD or PLAY only, not both at once.
6436 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6438 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6440 * gst/rtsp-server/rtsp-stream.c:
6441 rtsp-stream: RTCP and RTP transport cache cookies seperated
6442 RTCP packets were not sent because the same tr_cache_cookie was used for
6443 both RTP and RTCP. So only one of the tr_cache lists were populated
6444 depending on which one was sent first. If the tr_cache list is not
6445 populated then no packets can be sent. Most often this happened to be
6446 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6447 resulted in both the tr_cache_lists to be populated regardless of which
6449 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6451 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6453 * gst/rtsp-server/rtsp-stream.c:
6454 rtsp-stream: fix false compiler warning
6455 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6457 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6459 * gst/rtsp-server/rtsp-client.c:
6460 rtsp-client: log interleaved data received
6462 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6464 * gst/rtsp-server/rtsp-client.c:
6465 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6467 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6469 * gst/rtsp-server/rtsp-client.c:
6470 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6472 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6474 * gst/rtsp-server/rtsp-client.c:
6475 rtsp-client: Use a random session ID in the SDP
6476 RFC4566 Section 5.2 says that it should make the username, session id,
6477 nettype, addrtype and unicast address tuple globally unique. Always using
6478 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6479 Instead let's create a 64 bit random number, which at least brings us
6480 closer to the goal of global uniqueness.
6481 https://tools.ietf.org/html/rfc4566#section-5.2
6483 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6485 * examples/test-launch.c:
6486 * examples/test-mp4.c:
6487 * examples/test-ogg.c:
6488 * examples/test-uri.c:
6489 examples: Don't call gst_init() and gst_get_option_group()
6490 The latter calls the former at the appropriate time.
6492 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6494 * gst/rtsp-server/rtsp-client.c:
6495 rtsp-client: Drop trailing \0 of RTSP DATA messages
6496 We add a trailing \0 in GstRTSPConnection to make parsing of
6497 string message bodies easier (e.g. the SDP from DESCRIBE) but
6498 for actual data this means we have to drop it or otherwise
6499 create invalid data.
6501 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6503 * gst/rtsp-server/rtsp-stream.c:
6504 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6505 Fixes crash when two threads access handle_new_sample() at the same
6506 time, one for RTP, one for RTCP.
6507 Otherwise, when iterating over the transports cache, it might be modified by
6508 another thread at the same time if the transports cookie has changed.
6509 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6511 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6513 * gst/rtsp-server/rtsp-stream.c:
6514 rtsp-stream: Set format=TIME on our app sources for TCP
6516 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6518 * gst/rtsp-server/rtsp-session-pool.c:
6519 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6520 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6521 RFC 2326 states that session IDs may consist of alphanumeric as well as
6522 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6523 Previously the session ID was URI-escaped, this meant that any character
6524 which was not alphanumeric or any of the characters +-._~ would be
6525 percent encoded. While the RFC (surprisingly) mentions that linear white
6526 space in session IDs should be URI-escaped, it does not say anything
6527 about other characters. Moreover no white space is allowed in the
6528 session ID. Finally the percent character which is the result of
6529 URI-escaping is not allowed in a session ID.
6530 So there is no reason to do any URI-escaping, and now it is removed.
6531 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6533 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6536 Automatic update of common submodule
6537 From f2c6b95 to bc76a8b
6539 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6542 Fix 'make check' from top-level directory
6544 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6546 * examples/test-launch.c:
6547 * examples/test-mp4.c:
6548 * examples/test-ogg.c:
6549 * examples/test-uri.c:
6550 examples: Add command-line parsing and take a 'port' argument
6551 This allows users to run multiple servers on different ports for testing.
6552 Only done for examples that actually take arguments and hence are capable of
6553 outputting different streams for each instance on each port.
6554 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6556 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6558 * gst/rtsp-server/rtsp-client.c:
6559 * gst/rtsp-server/rtsp-client.h:
6560 rtsp-client: Add a send_message default signal handler
6561 This allows subclasses to easily hook into the response sending
6562 mechanism without doing everything from a signal, which seems
6563 awkward from subclasses.
6565 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6568 Automatic update of common submodule
6569 From ef1ffdc to f2c6b95
6571 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6575 configure: add --disable-examples switch
6576 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6578 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6580 * examples/.gitignore:
6581 * examples/Makefile.am:
6582 * examples/test-video-rtx.c:
6583 examples: add a retransmisison example implementing RFC4588
6584 Currently only SSRC-multiplexed rtx streams are supported
6586 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6588 * gst/rtsp-server/rtsp-stream.c:
6589 rtsp-stream: Fix some minor memory leaks
6591 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6593 * gst/rtsp-server/rtsp-media.c:
6594 rtsp-media: Some minor cleanup
6596 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6598 * gst/rtsp-server/rtsp-stream.c:
6599 rtsp-stream: Fix compiler warnings
6600 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6601 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6603 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6604 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6607 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6609 * docs/libs/gst-rtsp-server-sections.txt:
6610 * gst/rtsp-server/rtsp-media-factory.c:
6611 * gst/rtsp-server/rtsp-media-factory.h:
6612 * gst/rtsp-server/rtsp-media.c:
6613 * gst/rtsp-server/rtsp-media.h:
6614 * gst/rtsp-server/rtsp-sdp.c:
6615 * gst/rtsp-server/rtsp-stream.c:
6616 * gst/rtsp-server/rtsp-stream.h:
6617 media: implement ssrc-multiplexed retransmission support
6618 based off RFC 4588 and the server-rtpaux example in -good
6620 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6622 * gst/rtsp-server/rtsp-client.c:
6623 * gst/rtsp-server/rtsp-stream-transport.c:
6624 * gst/rtsp-server/rtsp-stream.c:
6625 rtsp: Ref transports in hash table.
6626 Also ref streams for transports.
6627 This solves a crash when reciving a rtcp after teardown but before
6629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6631 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6634 Automatic update of common submodule
6635 From 7bb2bce to ef1ffdc
6637 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6639 * gst/rtsp-server/rtsp-client.c:
6640 client: refactor cleanup of cached media
6642 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6644 * tests/check/gst/client.c:
6646 The session leak is now fixed, lets remove those FIXME comments.
6648 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6650 * tests/check/gst/rtspserver.c:
6651 tests: Test to setup two sessions on one connection
6652 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6654 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6656 * tests/check/gst/rtspserver.c:
6657 tests: Test setup with tcp transport
6658 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6660 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6662 * gst/rtsp-server/rtsp-client.c:
6663 client: Configure transport after creating session media
6664 The default implementation of configure_client_transport() in
6665 rtsp-client uses the session media when it chooses channels for
6666 interleaved traffic.
6667 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6669 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6671 * gst/rtsp-server/rtsp-client.c:
6672 * gst/rtsp-server/rtsp-session-media.c:
6673 client: Stop caching media in client when doing setup
6674 If the media has been managed by a session media, it should not be
6675 cached in the client any longer. The GstRTSPSessionMedia object is now
6676 responsible for unpreparing the GstRTSPMedia object using
6677 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6679 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6681 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6683 * gst/rtsp-server/rtsp-stream.c:
6684 rtsp-stream: unref srtp decoder when leaving bin
6685 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6687 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6689 * gst/rtsp-server/rtsp-client.c:
6690 rtsp-client: mikey memory leaks
6691 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6693 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6696 Automatic update of common submodule
6697 From 84d06cd to 7bb2bce
6699 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6702 Parallelise 'make check-valgrind'
6704 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6707 Automatic update of common submodule
6708 From a8c8939 to 84d06cd
6710 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6713 Automatic update of common submodule
6714 From 36388a1 to a8c8939
6716 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-media.c:
6719 rtsp-media: deactivate media when shutting down from paused
6720 This was only done when going directly from playing.
6721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
6723 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6725 * gst/rtsp-server/rtsp-client.c:
6726 * gst/rtsp-server/rtsp-context.h:
6727 rtsp-client: add stream transport to context
6728 We add the stream transport to the context so we can get the configured
6729 client stream transport in the setup request signal.
6730 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
6732 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6734 * gst/rtsp-server/rtsp-stream.c:
6735 stream: release lock even not all transports have been removed
6736 We don't want to keep the lock even we return FALSE because not all the
6737 transports have been removed. This could lead into a deadlock.
6738 https://bugzilla.gnome.org/show_bug.cgi?id=737797
6740 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
6742 * gst/rtsp-server/rtsp-sdp.c:
6743 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
6744 These were renamed in GstRTPBasePayload in 1.0
6746 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6748 * gst/rtsp-server/rtsp-client.c:
6749 client: set session media to NULL without the lock
6750 We need to set session medias to NULL without the client lock otherwise
6751 we can end up in a deadlock if another thread is waiting for the lock
6752 and media unprepare is also waiting for that thread to end.
6753 https://bugzilla.gnome.org/show_bug.cgi?id=737690
6755 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
6757 * gst/rtsp-server/rtsp-media.c:
6758 rtsp-media: Set state to UNPREPARING in all cases
6760 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
6762 * gst/rtsp-server/rtsp-media.c:
6763 media: set state to unpreparing when unprepare is initiated
6764 https://bugzilla.gnome.org/show_bug.cgi?id=737675
6766 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
6768 * gst/rtsp-server/rtsp-client.c:
6769 rtsp-client: Remove backlog limit while processings requests
6770 If the backlog limit is kept two cases of deadlocks may be
6771 encountered when streaming over TCP. Without the backlog
6772 limit this deadlocks can not happen, at the expence of
6774 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
6776 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
6778 * gst/rtsp-server/rtsp-client.c:
6779 rtsp-client: do not free main context before rtsp watch
6780 https://bugzilla.gnome.org/show_bug.cgi?id=737110
6782 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
6784 * tests/check/gst/rtspserver.c:
6785 tests: Extend unit test timeout to accomodate for valgrind
6786 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6788 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
6790 * gst/rtsp-server/rtsp-client.c:
6791 * gst/rtsp-server/rtsp-session.c:
6792 * gst/rtsp-server/rtsp-stream-transport.c:
6793 rtsp-*: Treat sending packets to clients as keepalive
6794 As long as gst-rtsp-server can successfully send RTP/RTCP data to
6795 clients then the client must be reading. This change makes the server
6796 timeout the connection if the client stops reading.
6797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6799 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
6801 * gst/rtsp-server/rtsp-client.c:
6802 rtsp-client: Allow backlog to grow while expiring session
6803 Allow the send backlog in the RTSP watch to grow to unlimited size while
6804 attempting to bring the media pipeline to NULL due to a session
6805 expiring. Without this change the appsink element cannot change state
6806 because it is blocked while rendering data in the new_sample callback.
6807 This callback will block until it has successfully put the data into the
6808 send backlog. There is a chance that the send backlog is full at this
6809 point which means that the callback may block for a long time, possibly
6810 forever. Therefore the media pipeline may also be prevented from
6811 changing state for a long time.
6812 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6814 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
6816 * gst/rtsp-server/rtsp-client.c:
6817 rtsp-client: Make old compilers happy
6818 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
6819 Just in case that guint8 doesn't fit in a pointer. Just in case ...
6821 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
6823 * gst/rtsp-server/rtsp-client.c:
6824 client: raise the backlog limits before pausing
6825 We need to raise the backlog limits before pausing the pipeline or else
6826 the appsink might be blocking in the render method in wait_backlog() and
6827 we would deadlock waiting for paused.
6828 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
6830 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
6832 * gst/rtsp-server/rtsp-client.c:
6833 client: make define for the WATCH_BACKLOG
6834 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
6836 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
6838 * gst/rtsp-server/rtsp-client.c:
6839 client: simplify session transport handling
6840 link/unlink of the transport in a session was done to keep track of all
6841 TCP transports and to send RTP/RTCP data to the streams. We can simplify
6842 that by putting all the TCP transports in a hashtable indexed with the
6844 We also don't need to link/unlink the transports when we pause/resume
6845 the streams. The same effect is already achieved when we pause/play the
6846 media. Indeed, when we pause the media, the transport is removed from
6847 the media and the callbacks will not be called anymore.
6848 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
6850 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
6852 * gst/rtsp-server/rtsp-stream-transport.c:
6853 * gst/rtsp-server/rtsp-stream-transport.h:
6854 stream-transport: make method to handle received data
6855 Make a method to handle the data received on a channel. It sends the
6856 data to the stream of the transport on the RTP or RTCP pads based on
6859 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
6861 * examples/test-mp4.c:
6862 test: add example of dumping RTCP reports
6864 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
6866 * gst/rtsp-server/rtsp-media.c:
6867 * gst/rtsp-server/rtsp-stream.c:
6868 * gst/rtsp-server/rtsp-stream.h:
6869 rtsp-media: Make sure that sequence numbers are monotonic after pause
6870 The sequence number is not monotonic for RTP packets after pause. The
6871 reason is basepayloader generates a randon sequence number when the
6872 pipeline goes from ready to pause. With this fix generation of sequence
6873 number will be monotonic when going from pause to play request.
6874 https://bugzilla.gnome.org/show_bug.cgi?id=736017
6876 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
6878 * gst/rtsp-server/rtsp-client.c:
6879 rtsp-client: Protect saved clients watch with a mutex
6880 Fixes a crash when close() is called while merging clients
6881 in handle_tunnel(). In that case close() would destroy the
6882 watch while it is still being used in handle_tunnel().
6883 https://bugzilla.gnome.org/show_bug.cgi?id=735570
6885 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
6887 * gst/rtsp-server/rtsp-stream.c:
6888 rtsp-stream: Remove the multicast group udp sources when removing from the bin
6890 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6892 * gst/rtsp-server/rtsp-media.c:
6893 * gst/rtsp-server/rtsp-stream.c:
6894 * gst/rtsp-server/rtsp-stream.h:
6895 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
6896 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
6897 seeking and will always continue counting the time. This leads to
6898 the NPT after a backwards seek to be something completely different
6899 to the actual seek position.
6900 https://bugzilla.gnome.org/show_bug.cgi?id=732644
6902 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
6904 * examples/test-appsrc.c:
6905 examples: fix another reference leak
6906 gst_rtsp_media_get_element() returns a new ref.
6908 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6910 * examples/test-appsrc.c:
6911 examples: unref element after usage
6912 gst_bin_get_by_name_recurse_up() returns an element
6913 reference that must be unreffed after usage.
6914 https://bugzilla.gnome.org/show_bug.cgi?id=734546
6916 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
6918 * gst/rtsp-server/rtsp-media.c:
6919 signals: Fix copy-pasto in target-state signal offset
6921 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
6925 Makefile: Add usage of build-checks step
6926 Allows building checks without running them
6928 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
6930 * gst/rtsp-server/rtsp-stream.c:
6931 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
6932 When a UDP multicast transport is used it is expected that the server listens
6933 for RTP and RTCP packets on the multicast group with the corresponding port.
6934 Without this we will never get RTCP packets from clients in multicast mode.
6935 https://bugzilla.gnome.org/show_bug.cgi?id=732238
6937 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6942 === release 1.4.0 ===
6944 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6950 * gst-rtsp-server.doap:
6953 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
6955 * gst/rtsp-server/rtsp-media.h:
6956 media: correct misspelled words in description
6957 https://bugzilla.gnome.org/show_bug.cgi?id=733244
6959 === release 1.3.91 ===
6961 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6967 * gst-rtsp-server.doap:
6970 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
6972 * docs/libs/gst-rtsp-server-sections.txt:
6975 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
6977 * gst/rtsp-server/rtsp-server.c:
6978 server: implement client REMOVE filter
6980 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
6982 * gst/rtsp-server/rtsp-client.c:
6983 * gst/rtsp-server/rtsp-client.h:
6984 client: expose _close() method
6985 Expose a previously internal close method to close the client
6988 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
6990 * gst/rtsp-server/rtsp-session-pool.c:
6991 session-pool: signal session-removed outside of the lock
6992 Release the lock before emiting the session-removed signal.
6994 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
6996 * gst/rtsp-server/rtsp-client.c:
6997 * gst/rtsp-server/rtsp-server.c:
6998 * gst/rtsp-server/rtsp-session-pool.c:
6999 * gst/rtsp-server/rtsp-session.c:
7000 * gst/rtsp-server/rtsp-stream.c:
7001 filter: Release lock in filter functions
7002 Release the object lock before calling the filter functions. We need to
7003 keep a cookie to detect when the list changed during the filter
7004 callback. We also keep a hashtable to make sure we only call the filter
7005 function once for each object in case of concurrent modification.
7006 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7008 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7010 * gst/rtsp-server/rtsp-client.c:
7011 client: check if watch is set in handle_teardown()
7012 The unit tests run without a watch
7014 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7016 * tests/check/gst/client.c:
7017 client tests: send teardown to cleanup session
7019 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7021 * tests/check/gst/rtspserver.c:
7022 server tests: send teardown to cleanup session
7024 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7026 * gst/rtsp-server/rtsp-client.c:
7027 client: keep ref to client for the session removed handler
7028 This extra ref will be dropped when all client sessions have been
7029 removed. A session is removed when a client sends teardown, closes its
7030 endpoint of the TCP connection or the sessions expires.
7031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7033 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7035 * gst/rtsp-server/rtsp-client.c:
7036 * gst/rtsp-server/rtsp-session.c:
7037 * tests/check/gst/client.c:
7038 client: manage media in session as a last step
7039 Once we manage a media in a session, we can't unmanage it anymore
7040 without destroying it. Therefore, first check everything before we
7041 manage the media, otherwise if something is wrong we have no way to
7043 If we created a new session and something went wrong, remove the session
7044 again. Fixes a leak in the unit test.
7046 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7048 * examples/test-mp4.c:
7049 * examples/test-ogg.c:
7050 examples: print 'stream ready at url' for mp4 and ogg example
7052 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7054 * gst/rtsp-server/rtsp-client.c:
7055 * gst/rtsp-server/rtsp-sdp.c:
7056 rtsp: fix for MIKEY api change
7058 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7060 * gst/rtsp-server/rtsp-client.c:
7061 client: free watch context only once
7062 The watch context is freed when the source is destroyed. Avoids
7063 a CRITICAL when we try to unref the context twice.
7065 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7067 * gst/rtsp-server/rtsp-client.c:
7070 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7072 * gst/rtsp-server/rtsp-client.c:
7073 client: protect sessions with lock
7074 Protect the list of sessions with the lock.
7075 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7077 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7079 * gst/rtsp-server/rtsp-client.c:
7080 Client: keep a ref to the session
7081 Don't just keep a weak ref to the session objects but use a hard ref. We
7082 will be notified when a session is removed from the pool (expired) with
7083 the new session-removed signal.
7084 Don't automatically close the RTSP connection when all the sessions of
7085 a client are removed, a client can continue to operate and it can create
7086 a new session if it wants. If you want to remove the client from the
7087 server, you have to use gst_rtsp_server_client_filter() now.
7088 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7089 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7091 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7093 * gst/rtsp-server/rtsp-session-pool.c:
7094 * gst/rtsp-server/rtsp-session-pool.h:
7095 session-pool: add session-removed signal
7096 Add a signal to be notified when a session is removed from the pool.
7098 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7100 * gst/rtsp-server/Makefile.am:
7101 * gst/rtsp-server/rtsp-server.h:
7102 Make rtsp-server.h a single-include header, use it for G-I
7103 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7105 === release 1.3.90 ===
7107 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7113 * gst-rtsp-server.doap:
7116 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7118 * gst/rtsp-server/rtsp-stream.c:
7119 stream: crypto can be NULL
7121 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7123 * gst/rtsp-server/rtsp-client.c:
7124 * gst/rtsp-server/rtsp-media.c:
7125 * gst/rtsp-server/rtsp-mount-points.c:
7126 introspection: add missing allow-none annotations
7127 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7129 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7131 * gst/rtsp-server/rtsp-address-pool.c:
7132 * gst/rtsp-server/rtsp-media.c:
7133 * gst/rtsp-server/rtsp-session-media.c:
7134 * gst/rtsp-server/rtsp-session-pool.c:
7135 * gst/rtsp-server/rtsp-stream-transport.c:
7136 * gst/rtsp-server/rtsp-stream.c:
7137 * gst/rtsp-server/rtsp-token.c:
7138 introspection: add (nullable) annotations to return values
7139 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7141 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7143 * gst/rtsp-server/rtsp-client.c:
7144 * gst/rtsp-server/rtsp-stream.c:
7145 gi: improve annotations
7146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7148 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7150 * gst/rtsp-server/rtsp-client.c:
7151 * gst/rtsp-server/rtsp-media-factory.c:
7152 * gst/rtsp-server/rtsp-media.c:
7153 * gst/rtsp-server/rtsp-server.c:
7154 signals: use generic marshal function
7155 Use the generic C marshal function.
7156 Use more explicit type instead of G_TYPE_POINTER
7158 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7160 * gst/rtsp-server/rtsp-context.h:
7161 context: add type macro
7163 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7165 * gst/rtsp-server/rtsp-client.c:
7166 * gst/rtsp-server/rtsp-sdp.c:
7167 * gst/rtsp-server/rtsp-sdp.h:
7168 sdp: hide key length defines
7169 They don't have a namespace.
7171 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7176 === release 1.3.3 ===
7178 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7184 * gst-rtsp-server.doap:
7187 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7189 * gst/rtsp-server/rtsp-client.c:
7190 * gst/rtsp-server/rtsp-sdp.c:
7191 * gst/rtsp-server/rtsp-sdp.h:
7192 mikey: add different key length parameters
7193 Add encryption and authentication key length parameters to MIKEY. For
7194 the encoders, the key lengths are obtained from the cipher and auth
7195 algorithms set in the caps. For the decoders, they are obtained while
7196 parsing the key management from the client.
7197 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7199 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7201 * tests/check/gst/stream.c:
7202 stream tests: Make sure we get right multicast address from stream
7203 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7205 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7207 * gst/rtsp-server/rtsp-client.c:
7208 client: ref the context until rtsp watch is alive
7209 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7211 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7213 * gst/rtsp-server/rtsp-client.c:
7214 client: Destroy the rtsp watch after connection close
7216 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7218 * gst/rtsp-server/rtsp-media.c:
7219 media: fix confusing comment
7221 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7223 * gst/rtsp-server/rtsp-session.c:
7224 rtsp-session: Timeout in header.
7225 Adding the possbilty to always have timout in header.
7226 This is configurabe with setting "timeout-always-visible".
7227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7229 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7234 === release 1.3.2 ===
7236 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7243 * gst-rtsp-server.doap:
7246 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7249 Automatic update of common submodule
7250 From 211fa5f to 1f5d3c3
7252 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7254 * gst/rtsp-server/rtsp-client.c:
7255 client: store TCP ports in transport
7256 Store the TCP ports in the transport when we are doing RTSP over TCP.
7257 This way, we can easily get to the ports from the transport.
7258 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7260 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7262 * gst/rtsp-server/rtsp-stream.c:
7263 stream: add signals for new RTP/RTCP encoders
7264 New signals to allow the user to configure the dynamically created
7266 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7268 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7270 * gst/rtsp-server/rtsp-media.c:
7271 * gst/rtsp-server/rtsp-media.h:
7272 media: Make suspend()/unsuspend() virtual
7273 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7275 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7277 * gst/rtsp-server/rtsp-client.c:
7278 client: fix send-message signal marshaller
7279 Use generic marshalling for the send-message signal. It has
7280 two POINTER arguments, not just one.
7281 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7283 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7285 * tests/check/gst/media.c:
7286 tests: add and remove pads only once
7287 In this test we simulate a dynamic pad by watching the caps event.
7288 Because of renegotiation in the base payloader now, this caps is sent
7289 multiple times but we can only deal with 1 invocation, use a variable to
7290 only 'add and remove' the pad once.
7292 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7294 * tests/check/gst/rtspserver.c:
7295 tests: add unit test for correct handling of Require headers
7296 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7298 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7300 * gst/rtsp-server/rtsp-client.c:
7301 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7302 Servers must handle Require headers and must report a failure
7303 if they don't handle any of the Required options, see RFC 2326,
7304 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7305 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7307 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7312 === release 1.3.1 ===
7314 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7320 * gst-rtsp-server.doap:
7323 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7326 Automatic update of common submodule
7327 From bcb1518 to 211fa5f
7329 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7334 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7336 * tests/check/gst/sessionmedia.c:
7337 tests: fix memory leak in sessionmedia unit test
7339 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7341 * gst/rtsp-server/rtsp-client.c:
7342 client: emit a signal before sending a message
7343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7345 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7347 * gst/rtsp-server/rtsp-client.c:
7348 client: pass context to send_message
7349 Pass the current context to send_message, we will need it later.
7351 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7353 * gst/rtsp-server/rtsp-client.c:
7354 client: fix typo in comment
7356 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7358 * gst/rtsp-server/rtsp-media.c:
7359 media: Do not stop thread twice if default_prepare() fails
7361 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7363 * gst/rtsp-server/rtsp-client.c:
7364 client: set the watch to flushing before going to NULL
7365 First set the watch to flushing so that we unblock any current and
7366 future attempt to send data on the watch, Then set the pipeline to
7368 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7370 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7372 * gst/rtsp-server/rtsp-session-pool.c:
7373 * tests/check/gst/sessionpool.c:
7374 rtsp-session-pool: Fixes annotation
7375 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7376 in the sessionpool test.
7377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7379 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7381 * gst/rtsp-server/rtsp-media.c:
7382 * gst/rtsp-server/rtsp-media.h:
7383 media: make media_prepare virtual
7384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7386 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7388 * gst/rtsp-server/rtsp-media.c:
7389 * tests/check/gst/media.c:
7390 media: stop the thread in more error cases
7392 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7394 * gst/rtsp-server/rtsp-media.c:
7395 * tests/check/gst/media.c:
7396 media: allow NULL as the thread
7397 Use the default context whan passing a NULL thread.
7399 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7401 * gst/rtsp-server/rtsp-client.c:
7402 rtsp-client: indent cleanup
7403 Coverity was moaning about unreachable code, and I think it was just
7404 confused by { being before the label. We'll see if it pops up again.
7407 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7409 * gst/rtsp-server/rtsp-client.c:
7410 * gst/rtsp-server/rtsp-media.c:
7411 client: Add drop-backlog property
7412 When we have too many messages queued for a client (currently hardcoded
7413 to 100) we overflow and drop the messages. Add a drop-backlog property
7414 to control this behaviour. Setting this property to FALSE will retry
7415 to send the messages to the client by waiting for more room in the
7417 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7419 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7421 * gst/rtsp-server/rtsp-client.c:
7422 client: support for POST before GET when setting up a tunnel
7424 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7426 * gst/rtsp-server/rtsp-client.c:
7427 client: remove watch of the second client after http tunnel setup
7428 The second client will be freed after the HTTP tunnel has been set up.
7429 Make sure it's RTSP watch is never dispatched again.
7430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7432 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7434 * gst/rtsp-server/rtsp-media.c:
7435 * tests/check/gst/media.c:
7436 media: Make media_prepare() fail if port allocation fails
7437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7439 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7441 * tests/check/gst/media.c:
7442 media test: cleanup the thread pool in tests
7444 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7446 * gst/rtsp-server/rtsp-media.c:
7447 * tests/check/gst/media.c:
7448 rtsp-media: Unblock blocked streams in unprepare
7449 The streams will be blocked when a live media is prepared.
7450 The streams should be unblocked in gst_rtsp_media_unprepare.
7451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7453 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7455 * gst/rtsp-server/rtsp-media.c:
7456 media: release the state lock when going to NULL
7457 Set our state to UNPREPARING and release the state-lock before
7458 setting the pipeline to the NULL state. This way, any pad-added
7459 callback will be able to take the state-lock and check that we are now
7460 unpreparing instead of deadlocking.
7461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7463 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7465 * gst/rtsp-server/rtsp-media.c:
7466 media: protect status with lock
7467 Make sure we only update the status with the lock.
7469 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7471 * gst/rtsp-server/rtsp-client.c:
7472 * gst/rtsp-server/rtsp-sdp.c:
7473 rtsp: update for MIKEY API changes
7475 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7477 * gst/rtsp-server/rtsp-client.c:
7478 client: parse the mikey response from the client
7479 Parse the mikey response from the client and update the policy for
7482 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7484 * gst/rtsp-server/rtsp-stream.c:
7485 * gst/rtsp-server/rtsp-stream.h:
7486 stream: add method to set crypto info
7487 Make a method to configure the crypto information of a stream.
7488 Set udpsrc in READY instead of PAUSED so that we can configure caps
7491 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7493 * gst/rtsp-server/rtsp-client.c:
7494 client: cleanup error paths
7496 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7498 * gst/rtsp-server/rtsp-media.c:
7501 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7503 * examples/test-video.c:
7504 test: enable SRTP only on RTSPS
7505 We only want to enable SRTP when doing rtsp over TLS so that we can
7506 exchange the keys in a secure way.
7508 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7510 * examples/test-video.c:
7511 test: print an error on failure
7513 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7516 * examples/test-video.c:
7517 * gst/rtsp-server/rtsp-sdp.c:
7518 * gst/rtsp-server/rtsp-stream.c:
7519 * tests/check/Makefile.am:
7520 stream: add SRTP support
7521 Install srtp encoder and decoder elements in rtpbin
7524 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7526 * tests/check/Makefile.am:
7527 * tests/check/gst/sessionpool.c:
7528 tests: Add unit tests for sessionpool
7529 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7531 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7533 * tests/check/gst/threadpool.c:
7534 tests: Improve code coverage of rtsp-threadpool tests
7535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7537 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7539 * tests/check/gst/sessionmedia.c:
7540 tests: Improve code coverage for rtsp-session-media
7541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7543 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7545 gobject-introspection: Add annotations to support language bindings
7546 In addition a few cosmetic changes:
7547 * Adjust the order of arguments
7548 * Fix typo: occured -> occurred
7549 * Fix indentation after Return:-clauses
7550 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7552 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7554 * gst/rtsp-server/rtsp-stream.c:
7555 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7556 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7558 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7560 * gst/rtsp-server/rtsp-stream.c:
7561 stream: take caps after the session manager
7562 Take the caps for the SDP after they leave the rtpbin so that we can
7563 also get the properties added by rtpbin elements.
7565 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7567 * gst/rtsp-server/rtsp-stream.c:
7568 stream: release lock while pushing out packets
7569 Keep a cache of the transports and use this to iterate the transport
7570 while pushing packets. This allows us to release the lock early.
7571 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7573 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7575 * gst/rtsp-server/rtsp-client.c:
7576 * gst/rtsp-server/rtsp-client.h:
7577 rtsp-client: vmethod for modifying tunnel GET response
7578 Add a vmethod tunnel_http_response where the response to the HTTP GET
7579 for tunneled connections can be modified.
7580 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7582 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7584 * gst/rtsp-server/rtsp-sdp.c:
7585 sdp: make 1 media line per profile
7586 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7587 line in the SDP for each profile. The client is then supposed to pick
7588 one of the profiles in the SETUP request. Because the m= lines have the
7589 same pt, the client also knows that only 1 option is possible.
7591 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7593 * gst/rtsp-server/rtsp-media-factory.c:
7594 * gst/rtsp-server/rtsp-media-factory.h:
7595 * gst/rtsp-server/rtsp-media.c:
7596 factory: add profile property and pass to media and streams
7598 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7600 * examples/test-multicast.c:
7601 * gst/rtsp-server/rtsp-sdp.c:
7602 sdp: pass multicast connection for multicast-only stream
7603 Pass the multicast address of the stream in the connection info in the
7604 SDP so that clients try a multicast connection first.
7605 Only allow multicast connections in the test-multicast example. Also
7606 increase the TTL a little.
7608 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7611 .gitignore: Ignore gcov intermediate files
7612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7614 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7616 * gst/rtsp-server/rtsp-stream.c:
7617 stream: release some locks in error cases
7619 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7621 docs: Enable and fix gtk-doc warnings
7622 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7623 * addresspool/mediafactory: Add missing annotation colon
7624 * stream: Annotate return value
7625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7627 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7630 Automatic update of common submodule
7631 From fe1672e to bcb1518
7633 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7636 Automatic update of common submodule
7637 From 1a07da9 to fe1672e
7639 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7641 * examples/Makefile.am:
7642 examples: use LDADD for libs instead of LDFLAGS
7644 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7647 configure: make sure releases are in .doap file
7649 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7651 * examples/test-cgroups.c:
7652 examples: test-cgroups: don't put code with side effects into g_assert()
7653 The g_assert() might get compiled out with the right
7654 compiler/preprocessor flags.
7656 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7658 * examples/.gitignore:
7659 examples: add cgroup test binary to .gitignore
7661 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7663 * examples/test-cgroups.c:
7664 examples: fix cgroup test build
7665 Fixes build failure caused by compiler warning:
7666 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7668 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7671 .gitignore: ignore temp files created in the course of 'make check'
7673 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7675 * gst/rtsp-server/rtsp-media.c:
7676 rtsp-media: don't loose frames handling new PLAY request
7677 If client supplied a range check if the range specifies the start point.
7678 If not, then do an accurate seek to the current position. If a start
7679 point was specified do do a key unit seek to make sure the streaming
7680 starts with decodeable frames.
7681 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7683 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7685 * gst/rtsp-server/rtsp-media.c:
7686 Revert "media: only flush when setting a new start position"
7687 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7688 We need to do the flush in all cases, demuxer block currently for
7691 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7693 * gst/rtsp-server/rtsp-media.c:
7694 media: only flush when setting a new start position
7695 Only flush the pipeline when we change the start position with
7697 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7699 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7701 * gst/rtsp-server/rtsp-stream.c:
7702 stream: set ttl-mc before adding the socket
7703 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7704 never be set on socket.
7705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7707 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7709 * gst/rtsp-server/rtsp-media.c:
7710 media: stop thread if media is already prepared
7711 in gst_rtsp_media_prepare() the thread is not used if media is already
7712 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7714 https://bugzilla.gnome.org/show_bug.cgi?id=724182
7716 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
7719 build: Ship gst-rtsp-server.doap file
7721 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
7723 * tests/check/gst/rtspserver.c:
7724 tests: Fix another compiler warning with gcc
7726 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
7728 * gst/rtsp-server/rtsp-client.c:
7729 * gst/rtsp-server/rtsp-mount-points.c:
7730 * gst/rtsp-server/rtsp-stream.c:
7731 * tests/check/gst/client.c:
7732 rtsp-server: Fix lots of compiler warnings with clang
7734 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
7737 * gst-rtsp-server.doap:
7738 * tests/Makefile.am:
7739 configure: Synchronise with the configure scripts of the other modules
7741 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
7744 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
7746 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
7748 * gst/rtsp-server/rtsp-media.c:
7749 * gst/rtsp-server/rtsp-stream.c:
7750 Revert "rtsp-server: support build against last stable release"
7751 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
7752 Let us require 1.2.3 now, which is going to be released in a few
7755 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
7757 * gst/rtsp-server/rtsp-session-media.c:
7758 * gst/rtsp-server/rtsp-stream-transport.c:
7759 session: improve RTP-Info
7760 Ignore streams that can't generate RTP-Info instead of failing.
7761 Don't return the empty string when all streams are unconfigured but
7762 return NULL so that we don't generate and empty RTP-Info header.
7763 Improve docs a little.
7765 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
7767 * gst/rtsp-server/rtsp-session-media.c:
7768 Don't free rtpinfo GString when it is NULL
7769 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7771 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
7773 * gst/rtsp-server/rtsp-media.c:
7774 media: only set keyframe flag when modifying start
7775 Only set the keyframe flag when we modify the start position. The
7776 keyframe flag should probably be ignored when no change is requested but
7777 until we can claim this is all documented properly and all demuxer
7778 implement this, avoid setting the flag.
7779 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
7781 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
7783 * gst/rtsp-server/rtsp-thread-pool.c:
7784 thread-pool: Unref source after mainloop has quit to avoid races in GLib
7785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
7787 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
7789 * gst/rtsp-server/rtsp-stream.c:
7790 stream: handle NULL seqnum and rtptime arguments
7792 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
7794 * gst/rtsp-server/rtsp-thread-pool.c:
7795 * tests/check/gst/threadpool.c:
7796 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
7797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
7799 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
7801 * gst/rtsp-server/rtsp-stream.c:
7802 stream: add fallback for missing stats property
7803 Use a fallback when the payloader does not have a stats property
7804 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7806 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
7809 Automatic update of common submodule
7810 From f7bc1c3 to 1a07da9
7812 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
7814 * gst/rtsp-server/rtsp-stream.c:
7815 stream: don't leak stats structure
7816 Don't leak the stats structure and deal with NULL stats.
7818 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
7820 * gst/rtsp-server/rtsp-stream.c:
7821 stream: Get rtpinfo properties atomically from payloader
7822 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
7824 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
7826 * gst/rtsp-server/rtsp-media.c:
7827 media: refactor state change functions and signals
7828 Make functions to set the target state and the pipeline state and emit
7829 the signals from those functions.
7831 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
7833 * gst/rtsp-server/rtsp-media.c:
7834 * gst/rtsp-server/rtsp-media.h:
7835 media: add signal to notify of pending state changes
7837 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
7839 * gst/rtsp-server/rtsp-media.c:
7840 * gst/rtsp-server/rtsp-stream.c:
7841 rtsp-server: support build against last stable release
7842 Until 1.2.3 is out with the new get_type function and we
7845 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
7847 * gst/rtsp-server/rtsp-stream.c:
7848 stream: fix compilation
7850 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
7852 * gst/rtsp-server/rtsp-media.c:
7853 * gst/rtsp-server/rtsp-media.h:
7854 * gst/rtsp-server/rtsp-stream.c:
7855 * gst/rtsp-server/rtsp-stream.h:
7856 stream: add property to configure profiles
7858 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
7860 * gst/rtsp-server/rtsp-client.c:
7861 client: let stream check supported transport
7862 Delegate the check if a transport is allowed to the stream.
7863 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
7865 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
7867 * gst/rtsp-server/rtsp-stream.c:
7868 * gst/rtsp-server/rtsp-stream.h:
7869 stream: add method to check supported transport
7870 Add a method to check if a transport is supported
7872 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
7875 configure.ac: Only check for gstreamer-check, not check
7876 We include check in gstreamer-check since quite some time now.
7878 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
7880 * gst/rtsp-server/rtsp-session-media.c:
7881 * gst/rtsp-server/rtsp-stream-transport.c:
7882 * gst/rtsp-server/rtsp-stream.c:
7883 * gst/rtsp-server/rtsp-stream.h:
7884 stream: return clock-rate from get_rtpinfo
7885 And use it to correct the rtptime to the requested start-time.
7886 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
7888 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
7890 * gst/rtsp-server/rtsp-session-media.c:
7891 * gst/rtsp-server/rtsp-stream-transport.c:
7892 * gst/rtsp-server/rtsp-stream-transport.h:
7893 session-media: calculate start-time
7895 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
7897 * gst/rtsp-server/rtsp-stream-transport.c:
7898 * gst/rtsp-server/rtsp-stream.c:
7899 * gst/rtsp-server/rtsp-stream.h:
7900 stream: also return the running-time
7901 Return the running-time in the rtpinfo as well.
7903 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
7905 * gst/rtsp-server/rtsp-client.c:
7906 * gst/rtsp-server/rtsp-session-media.c:
7907 * gst/rtsp-server/rtsp-session-media.h:
7908 * gst/rtsp-server/rtsp-stream-transport.c:
7909 * gst/rtsp-server/rtsp-stream-transport.h:
7910 session-media: let the session-media make the RTPInfo
7911 Add method to create the RTPInfo for a stream-transport.
7912 Add method to create the RTPInfo for all stream-transports in a
7914 Use the session-media RTPInfo code in client. This allows us to refactor
7915 another method to link the TCP callbacks.
7917 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7919 mount-points: sort sequence before g_sequence_lookup
7920 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
7921 sort sequence if dirty, otherwise lookup will fail.
7922 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
7924 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
7927 configure: rename package from gst-rtsp to gst-rtsp-server
7928 To match git module name and avoid confusion with the
7929 rtsp lib in gst-plugins-base and rtsp plugin in -good.
7931 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
7934 configure: bump core/base/good requirement to 1.2.0
7935 Bump to released stable version and make implicit
7936 requirements explicit.
7938 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
7943 Fix broken gettext setup which is not used anyway
7945 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
7948 Automatic update of common submodule
7949 From dbedaa0 to d48bed3
7951 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
7953 * gst/rtsp-server/rtsp-client.c:
7954 * gst/rtsp-server/rtsp-media.c:
7955 * gst/rtsp-server/rtsp-media.h:
7956 media: add setup_sdp vmethod
7957 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
7958 gst_rtsp_media_setup_sdp.
7959 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
7961 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
7963 * gst/rtsp-server/rtsp-stream.c:
7964 rtsp-stream: Check return value of sscanf
7965 streamid is only valid if sscanf matched something.
7967 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
7969 * gst/rtsp-server/rtsp-client.c:
7970 rtsp-client: Fix iteration
7971 Wouldn't even enter the code block otherwise (i++ was used as the check
7972 and not the postfix).
7974 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
7976 * gst/rtsp-server/rtsp-client.c:
7977 * gst/rtsp-server/rtsp-client.h:
7978 client: add vmethod to configure media and streams
7979 Implement a vmethod that can be used to configure the media and the
7980 streams based on the current context. Handle the blocksize handling in
7981 the default handler.
7982 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
7984 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7987 Make git ignore more unit test binaries
7989 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7991 * gst/rtsp-server/rtsp-address-pool.h:
7992 * gst/rtsp-server/rtsp-auth.h:
7993 * gst/rtsp-server/rtsp-client.h:
7994 * gst/rtsp-server/rtsp-context.h:
7995 * gst/rtsp-server/rtsp-media-factory-uri.h:
7996 * gst/rtsp-server/rtsp-media-factory.h:
7997 * gst/rtsp-server/rtsp-media.h:
7998 * gst/rtsp-server/rtsp-mount-points.h:
7999 * gst/rtsp-server/rtsp-server.h:
8000 * gst/rtsp-server/rtsp-session-media.h:
8001 * gst/rtsp-server/rtsp-session-pool.h:
8002 * gst/rtsp-server/rtsp-session.h:
8003 * gst/rtsp-server/rtsp-stream-transport.h:
8004 * gst/rtsp-server/rtsp-stream.h:
8005 * gst/rtsp-server/rtsp-thread-pool.h:
8006 * gst/rtsp-server/rtsp-token.h:
8007 rtsp-server: add padding to many public structures
8008 Not mini objects though, since they are not subclassable
8009 anyway, nor kept on the stack or inlined in a structure.
8011 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8013 media: add new create_rtpbin vmethod
8014 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8015 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8017 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8019 * tests/check/gst/media.c:
8020 tests: fix memory leak, free test's thread pool
8021 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8023 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8025 * gst/rtsp-server/rtsp-stream-transport.c:
8026 stream-transport: free url in finalize
8028 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8030 * gst/rtsp-server/rtsp-media.c:
8031 media: also do state change in suspended state
8033 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8035 * gst/rtsp-server/rtsp-client.c:
8036 * gst/rtsp-server/rtsp-media.c:
8037 media: also handle prepare and range in suspended state
8038 When we are suspended, we are already prepared.
8039 We can get the range in the suspended state.
8041 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8043 * tests/check/Makefile.am:
8044 * tests/check/gst/sessionmedia.c:
8045 check: add test for uri in setup
8046 Added unit tests for the new functionality in GstRTSPStreamTransport.
8047 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8049 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8051 * gst/rtsp-server/rtsp-client.c:
8052 client: store setup uri and use in PLAY response
8053 Store the uri used when doing the setup and use that in the PLAY
8055 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8057 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8059 * gst/rtsp-server/rtsp-stream-transport.c:
8060 * gst/rtsp-server/rtsp-stream-transport.h:
8061 stream-transport: add method to get/set url
8063 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8065 * gst/rtsp-server/rtsp-client.c:
8066 client: suspend after SDP and unsuspend before PLAYING
8067 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8068 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8070 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8072 * gst/rtsp-server/rtsp-media-factory.c:
8073 * gst/rtsp-server/rtsp-media-factory.h:
8074 * gst/rtsp-server/rtsp-media.c:
8075 * gst/rtsp-server/rtsp-media.h:
8076 * gst/rtsp-server/rtsp-session-media.c:
8077 * gst/rtsp-server/rtsp-session.c:
8078 * tests/check/gst/media.c:
8079 * tests/check/gst/mediafactory.c:
8080 media: add suspend modes
8081 Add support for different suspend modes. The stream is suspended right after
8082 producing the SDP and after PAUSE. Different suspend modes are available that
8083 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8084 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8085 state and RESET will bring the pipeline to the NULL state.
8086 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8087 this means that the pipeline needs to be prerolled again.
8088 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8089 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8091 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8093 * gst/rtsp-server/rtsp-media.c:
8094 media: start live streams in blocked state
8095 Start live streams in the blocked state and make them preroll using the
8096 messages. This ensure that no data is played by the sink until we explicitly
8097 unblock the stream right before going to PLAYING.
8098 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8100 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8102 * gst/rtsp-server/rtsp-media.c:
8103 media: refactor starting and waiting for preroll
8104 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8105 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8107 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8109 * gst/rtsp-server/rtsp-stream.c:
8110 * gst/rtsp-server/rtsp-stream.h:
8111 stream: add API to block streams
8112 Add an API to block on the streams and make it post a message.
8113 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8114 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8116 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8118 * docs/libs/Makefile.am:
8119 docs: Specify the override file
8120 Even if it's empty (for now) it avoids make distcheck complaining
8122 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8124 * gst/rtsp-server/rtsp-media.c:
8125 media: move default implementations to where they are used
8127 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8129 * gst/rtsp-server/rtsp-media.c:
8130 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8131 We need to take the state_lock when calling this method.
8133 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8135 * gst/rtsp-server/rtsp-media.c:
8136 media: handle add-added on non-bins too
8137 Handle dynamic payloaders that are not bins, as used in the unit-test.
8139 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8141 * gst/rtsp-server/rtsp-media-factory.c:
8142 * gst/rtsp-server/rtsp-media-factory.h:
8143 * gst/rtsp-server/rtsp-media.c:
8144 rtsp-media/-factory: Fix request pad name comments
8145 These must be escaped for gtk-doc to parse the comments without warnings.
8147 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8149 rtsp-media: remove transports if media is in error status
8150 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8151 trying to change to GST_STATE_NULL and media is in error status, we
8152 remove all transports.
8153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8155 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8157 * gst/rtsp-server/rtsp-media.c:
8158 rtsp-media: use element metadata to find payloader
8159 Use the element metadata to find the payloader instead of checking
8161 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8163 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8165 rtsp-stream: add getter for payload type
8166 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8167 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8168 element and create the stream with this one instead of the dynpay%d
8170 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8172 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8174 * gst/rtsp-server/rtsp-client.c:
8175 * gst/rtsp-server/rtsp-context.h:
8176 * gst/rtsp-server/rtsp-media.c:
8177 * gst/rtsp-server/rtsp-mount-points.c:
8178 * gst/rtsp-server/rtsp-server.c:
8179 * gst/rtsp-server/rtsp-token.c:
8180 rtsp-*: Refer to NULL as a constant in comments
8182 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8184 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8186 rtsp-*: Fix type name typos in comments
8187 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8188 * rtsp-auth: Refer to part of constant name as text
8189 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8190 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8191 * rtsp-stream: Fix typo when refering to GstBin
8192 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8194 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8197 * docs/libs/gst-rtsp-server-docs.sgml:
8198 * docs/libs/gst-rtsp-server-sections.txt:
8199 docs: Improve documentation
8200 * Include annotation-glossary to quiet gtk-doc
8201 * Rename remaining ClientState -> Context
8202 * Rename object hierarchy file
8203 * Remove stale chapter references
8204 * Add missing function and object references
8205 * Include missing GstRTSPAddressPoolResult
8206 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8208 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8210 * gst/rtsp-server/rtsp-client.c:
8211 * gst/rtsp-server/rtsp-server.c:
8212 * gst/rtsp-server/rtsp-session-pool.c:
8213 * gst/rtsp-server/rtsp-session.c:
8214 * gst/rtsp-server/rtsp-stream.c:
8215 rtsp-server: sprinkle some allow-none annotations for g-i
8217 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8219 * gst/rtsp-server/rtsp-stream.c:
8220 * gst/rtsp-server/rtsp-stream.h:
8221 stream: add method to filter transports
8222 Add a method to safely iterate and collect the stream transports
8223 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8225 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8227 * gst/rtsp-server/rtsp-client.c:
8228 * gst/rtsp-server/rtsp-server.c:
8229 * gst/rtsp-server/rtsp-session-pool.c:
8230 * gst/rtsp-server/rtsp-session.c:
8231 rtsp: allow NULL func in filters
8232 Passing a null function make the filters return a list of
8235 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8237 * gst/rtsp-server/rtsp-address-pool.c:
8238 * tests/check/gst/addresspool.c:
8239 address-pool: fix address increment
8240 Use a guint instead of guint8 to increment the address. It's still not
8241 completely correct because a guint might not be able to hold the complete
8242 address range, but that's an enhacement for later.
8243 Add unit test to test improved behaviour.
8244 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8246 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8248 * gst/rtsp-server/rtsp-client.c:
8249 * tests/check/gst/client.c:
8250 client: allow absolute path in requests
8251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8253 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8255 * gst/rtsp-server/rtsp-client.c:
8256 * gst/rtsp-server/rtsp-client.h:
8257 client: make make_path_from_uri a vmethod
8259 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8261 * docs/libs/gst-rtsp-server-sections.txt:
8262 * gst/rtsp-server/rtsp-stream.c:
8263 * gst/rtsp-server/rtsp-stream.h:
8264 * tests/check/Makefile.am:
8265 * tests/check/gst/stream.c:
8266 stream: Add functions to get rtp and rtcp sockets
8267 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8269 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8271 * gst/rtsp-server/rtsp-context.c:
8272 * gst/rtsp-server/rtsp-context.h:
8273 context: defing a GType for the context
8274 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8276 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8278 * gst/rtsp-server/Makefile.am:
8279 * gst/rtsp-server/rtsp-auth.c:
8280 * gst/rtsp-server/rtsp-context.c:
8281 * gst/rtsp-server/rtsp-media.c:
8282 * gst/rtsp-server/rtsp-mount-points.c:
8283 * gst/rtsp-server/rtsp-server.h:
8284 * gst/rtsp-server/rtsp-session-media.c:
8285 * gst/rtsp-server/rtsp-session.c:
8286 * gst/rtsp-server/rtsp-stream.c:
8287 Fixed several GIR warnings
8289 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8291 * gst/rtsp-server/rtsp-auth.c:
8294 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8296 * tests/check/Makefile.am:
8297 * tests/check/gst/token.c:
8298 tests: Add unit tests for token
8299 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8301 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8303 * gst/rtsp-server/rtsp-token.c:
8304 token: Validate args for gst_rtsp_token_is_allowed
8305 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8307 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8309 * gst/rtsp-server/rtsp-token.c:
8310 token: Fix bug when creating empty token
8311 We always want to have a valid GstStructure in the token.
8312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8314 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8316 * gst/rtsp-server/rtsp-thread-pool.c:
8317 thread-pool: avoid race in shutdown
8318 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8319 don't actually stop the mainloop ever. Solve this race by adding an idle source
8320 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8321 if quit was called before we started it.
8323 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8325 * tests/check/Makefile.am:
8326 * tests/check/gst/permissions.c:
8327 tests: Add unit tests for permissions
8328 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8330 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8332 * tests/check/gst/mediafactory.c:
8333 tests: Test mediafactory permissions
8334 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8336 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8338 * gst/rtsp-server/rtsp-permissions.c:
8339 permissions: Fix refcounting when adding/removing roles
8340 Previously a role that was removed was unreffed twice, and when
8341 replacing an existing role the replaced role was freed while still being
8342 referenced. Both bugs are now fixed.
8343 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8345 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8347 * tests/check/gst/media.c:
8348 * tests/check/gst/mediafactory.c:
8349 * tests/check/gst/rtspserver.c:
8350 tests: Check gst_rtsp_url_parse return value
8351 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8353 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8356 Automatic update of common submodule
8357 From 865aa20 to dbedaa0
8359 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8361 * gst/rtsp-server/rtsp-server.c:
8362 rtsp-server: Fix socket leak
8363 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8365 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8367 * gst/rtsp-server/rtsp-session-pool.c:
8368 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8369 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8371 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8373 * examples/.gitignore:
8374 * examples/test-video.c:
8375 examples: fix compilation when WITH_AUTH is defined
8376 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8378 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8381 gitignore: Add new test binary
8383 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8385 * tests/check/Makefile.am:
8386 * tests/check/gst/threadpool.c:
8387 thread-pool: Add unit test for the thread pools
8388 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8390 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8392 * gst/rtsp-server/rtsp-thread-pool.c:
8393 thread-pool: Fix thread leak when reusing threads
8394 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8396 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8398 * gst/rtsp-server/rtsp-server.c:
8399 * tests/check/gst/rtspserver.c:
8400 tests: fixed racy behavior in rtspserver tests
8401 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8403 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8405 * tests/check/gst/addresspool.c:
8406 tests: Improve address pool unit tests
8407 Add a range with mixed IPV4 and IPV6 addresses to pool.
8408 Get an IPV4 address from an IPV6-only pool.
8409 Get an IPV6 address from an IPV4-only pool.
8410 Reserve a IPV6 address from an IPV4-only pool.
8411 Check for unicast addresses in multicast-only pool.
8412 Check for unicast addresses in uni-/multicast-mixed pool.
8413 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8415 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8417 * gst/rtsp-server/rtsp-client.c:
8418 client: append query string in PAUSE/PLAY/TEARDOWN as well
8420 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8422 * gst/rtsp-server/rtsp-client.c:
8423 client: Add query to control path
8424 If the SETUP url contains a query it must be appended to the control
8425 path so that it matches any already created stream in the media. The
8426 query will also be appended to the session media path.
8428 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8430 * gst/rtsp-server/rtsp-media.c:
8431 rtsp-media: remove old line
8433 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8435 * gst/rtsp-server/rtsp-stream.c:
8436 stream: Correct control comparison
8437 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8439 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8441 * gst/rtsp-server/rtsp-media.c:
8442 media: Check dynamically if the pipeline supports seeking
8443 We should not depend on whether or not the pipeline state change
8444 returned NO_PREROLL or not. A media could dynamically change its
8445 element and switch from seekable to non seekable so it's best to test
8446 the seekable nature of the pipeline dynamically when we try to do a seek.
8448 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8450 * gst/rtsp-server/rtsp-media.c:
8451 media: Return FALSE if seeking is not supported
8453 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8455 * gst/rtsp-server/rtsp-media.c:
8456 rtsp-media: don't seek accurate by default
8457 Accurate seeking is perhaps a little overkill in the most common situation and
8458 causes some formats (mp3) over slow media to seek extremely slowly.
8460 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8462 * tests/check/gst/rtspserver.c:
8463 tests: fix unit test
8464 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8466 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8468 * gst/rtsp-server/rtsp-client.c:
8469 client: Reply 400 if media cannot be constructed
8470 Reply 400 Bad Request instead of 503 Service Unavailable if media
8471 cannot be constructed in SETUP.
8472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8474 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8476 * gst/rtsp-server/rtsp-client.c:
8477 client: Send setup reply once only
8478 If find_media() failed in handle_setup_request() two replies was sent.
8479 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8481 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8484 Automatic update of common submodule
8485 From 6b03ba7 to 865aa20
8487 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8489 * gst/rtsp-server/rtsp-server.c:
8490 server: Emit client-connected signal earlier
8491 Emit client-connected before the client ref is given to a GSource,
8492 otherwise client-connected can be emitted after the client object has
8495 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8497 * gst/rtsp-server/rtsp-address-pool.c:
8498 * gst/rtsp-server/rtsp-address-pool.h:
8499 * gst/rtsp-server/rtsp-stream.c:
8500 * tests/check/gst/addresspool.c:
8501 addresspool: return reason of failure
8502 Let gst_rtsp_address_pool_reserve_address() return the reason why
8503 the address could not be reserved.
8504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8506 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8509 autogen.sh: Sync behaviour with other GStreamer modules
8510 Allows building from outside of tree amongst other things
8512 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8515 Automatic update of common submodule
8516 From b613661 to 6b03ba7
8518 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8521 Automatic update of common submodule
8522 From 74a6857 to b613661
8524 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8527 Automatic update of common submodule
8528 From 01a7a46 to 74a6857
8530 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8532 * gst/rtsp-server/rtsp-client.c:
8533 client: Do not read beyond end of path string
8534 If the setup was done without a control url, make sure we don't try to read the
8535 non-existing control string and crash.
8537 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8539 * gst/rtsp-server/rtsp-client.c:
8540 client: Fix RTPInfo header
8541 Refactor the method to make the content_base.
8542 Use the content-base and the control url to construct the RTPInfo
8545 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8547 * gst/rtsp-server/rtsp-client.c:
8548 client: map url to path only in describe
8549 Only map the request url to a path in the DESCRIBE method. The SDP then
8550 contains the base and control urls that should be used to SETUP/PAUSE/
8551 PLAY/TEARDOWN the media.
8553 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * gst/rtsp-server/rtsp-client.c:
8556 Revert "client: map URL to path in requests"
8557 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8558 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8559 contains the base and control urls which are used in the SETUP, PLAY,
8560 PAUSE and TEARDOWN requests.
8562 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8564 * gst/rtsp-server/rtsp-client.c:
8565 client: map URL to path in requests
8567 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * gst/rtsp-server/rtsp-client.c:
8570 * gst/rtsp-server/rtsp-mount-points.c:
8571 * gst/rtsp-server/rtsp-mount-points.h:
8572 mount-points: make vmethod to make path from uri
8573 Make a vmethod to transform an url into a path. The path is then used to lookup
8574 the factory. This makes it possible to also use other bits of the url, such as
8575 the query parameters, to locate the factory.
8577 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8579 * gst/rtsp-server/rtsp-thread-pool.c:
8580 * gst/rtsp-server/rtsp-thread-pool.h:
8581 thread-pool: Add cleanup to wait for the threadpool to finish
8582 Also fix race condition if two threads are asking for the first
8583 thread from the thread pool at once. This would case two internal
8584 GThreadPools to be created.
8585 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8587 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8589 * gst/rtsp-server/rtsp-client.c:
8590 * tests/check/gst/client.c:
8591 client: free threadpool
8592 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8594 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8596 * tests/check/gst/mountpoints.c:
8597 mountpoints tests: unref matched factories
8598 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8600 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8602 * tests/check/gst/media.c:
8603 media tests: unref thread pool and caps
8604 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8606 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8608 * gst/rtsp-server/rtsp-auth.c:
8609 * gst/rtsp-server/rtsp-media-factory.c:
8610 * gst/rtsp-server/rtsp-media.c:
8611 auth, media, media-factory: unref permissions
8612 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8614 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * examples/Makefile.am:
8617 Makefile: add rule for appsrc example
8619 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8621 * examples/test-appsrc.c:
8622 tests: add appsrc example
8623 Add an example on how to use appsrc to feed the server pipeline with data.
8625 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8627 * gst/rtsp-server/rtsp-client.c:
8628 rtsp-client: remove query part from content-base string
8629 Make sure that after the control url has been resolved, it's
8630 not a part of the query-string.
8631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8633 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8635 * gst/rtsp-server/rtsp-client.c:
8636 client: don't check url in response
8637 There is no url or method in the response to check
8639 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8641 * gst/rtsp-server/rtsp-client.c:
8642 * gst/rtsp-server/rtsp-client.h:
8643 Add handle-response signal for when we receive a GET_PARAMETER response
8645 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8647 * gst/rtsp-server/rtsp-server.c:
8648 Fix gst_rtsp_server_client_filter, using wrong variable type
8650 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8652 * gst/rtsp-server/rtsp-media-factory-uri.c:
8653 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8654 For AAC we need to check for framed=true instead of parsed=true.
8655 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8657 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * gst/rtsp-server/rtsp-stream.c:
8660 stream: optimize pipeline for protocols
8661 When TCP is not an allowed protocol for the stream, avoid creating the
8662 appsrc/appsink/queue and tee elements.
8664 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8666 * gst/rtsp-server/rtsp-media.c:
8667 media: set protocols on streams
8669 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8671 * gst/rtsp-server/rtsp-client.c:
8672 client: use protocols supported by stream
8674 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8676 * gst/rtsp-server/rtsp-media-factory.c:
8677 * gst/rtsp-server/rtsp-media.c:
8678 * gst/rtsp-server/rtsp-stream.c:
8679 media-factory: allow all protocols
8681 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8683 * gst/rtsp-server/rtsp-media.c:
8684 media: configure protocols in new streams
8686 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8688 * gst/rtsp-server/rtsp-stream.c:
8689 * gst/rtsp-server/rtsp-stream.h:
8690 stream: add protocols property
8692 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8694 * gst/rtsp-server/rtsp-media.c:
8695 rtsp-media: send state in "new-state" signal
8696 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8698 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8701 build: add subdir-objects to AM_INIT_AUTOMAKE
8702 Fixes warnings with automake 1.14
8703 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8705 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8707 * docs/libs/gst-rtsp-server-sections.txt:
8708 * gst/rtsp-server/rtsp-client.c:
8709 * gst/rtsp-server/rtsp-server.c:
8710 * gst/rtsp-server/rtsp-server.h:
8711 server: add method to iterate clients of server
8713 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8715 * gst/rtsp-server/rtsp-media.c:
8716 * gst/rtsp-server/rtsp-media.h:
8717 Add vmethod for rtsp-media subclass to access rtpbin
8719 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8721 * gst/rtsp-server/rtsp-client.h:
8722 small documentation fix
8724 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8726 * gst/rtsp-server/rtsp-client.c:
8727 Do not take range header if range is invalid
8729 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8731 * docs/libs/gst-rtsp-server-sections.txt:
8732 * gst/rtsp-server/rtsp-media.c:
8733 media: add docs for new method
8735 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8737 * gst/rtsp-server/rtsp-media.c:
8738 * gst/rtsp-server/rtsp-media.h:
8739 Add API to rtsp-media set the pipeline's state
8741 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-media.c:
8744 Update current position/duration when gst_rtsp_media_get_range_string is called
8746 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8748 * examples/test-cgroups.c:
8749 tests: add some more docs
8751 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8753 * examples/test-cgroups.c:
8754 * gst/rtsp-server/Makefile.am:
8755 * gst/rtsp-server/rtsp-auth.c:
8756 * gst/rtsp-server/rtsp-auth.h:
8757 * gst/rtsp-server/rtsp-client.c:
8758 * gst/rtsp-server/rtsp-client.h:
8759 * gst/rtsp-server/rtsp-context.c:
8760 * gst/rtsp-server/rtsp-context.h:
8761 * gst/rtsp-server/rtsp-params.c:
8762 * gst/rtsp-server/rtsp-params.h:
8763 * gst/rtsp-server/rtsp-server.c:
8764 * gst/rtsp-server/rtsp-thread-pool.c:
8765 * gst/rtsp-server/rtsp-thread-pool.h:
8766 * tests/check/gst/client.c:
8767 ClientState -> Context
8768 Rename the clientstate to context and put the code in a separate file.
8770 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8772 * examples/test-auth.c:
8773 * gst/rtsp-server/rtsp-auth.c:
8774 * gst/rtsp-server/rtsp-auth.h:
8775 auth: add support for default token
8776 The default token is used when the user is not authenticated and can be used to
8777 give minimal permissions.
8779 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8781 * examples/test-auth.c:
8782 * gst/rtsp-server/rtsp-auth.c:
8783 auth: use defines when possible
8785 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8787 * gst/rtsp-server/rtsp-address-pool.c:
8788 address-pool: improve docs
8790 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8792 * gst/rtsp-server/rtsp-permissions.c:
8793 permissions: add the role to the copy
8795 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
8797 * gst/rtsp-server/rtsp-permissions.c:
8798 permissions: Also copy the roles
8800 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
8802 * gst/rtsp-server/rtsp-permissions.c:
8803 permissions: Make it build
8805 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8807 * gst/rtsp-server/rtsp-address-pool.h:
8810 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8812 * docs/libs/gst-rtsp-server-sections.txt:
8813 * gst/rtsp-server/rtsp-auth.c:
8814 * gst/rtsp-server/rtsp-auth.h:
8815 * gst/rtsp-server/rtsp-media.c:
8816 * gst/rtsp-server/rtsp-session-media.c:
8817 * gst/rtsp-server/rtsp-stream-transport.c:
8818 * gst/rtsp-server/rtsp-stream-transport.h:
8819 * gst/rtsp-server/rtsp-stream.c:
8820 * tests/check/gst/client.c:
8823 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8825 * docs/libs/gst-rtsp-server-sections.txt:
8826 * gst/rtsp-server/rtsp-address-pool.c:
8827 * gst/rtsp-server/rtsp-address-pool.h:
8828 * tests/check/gst/addresspool.c:
8829 * tests/check/gst/rtspserver.c:
8830 address-pool: cleanups
8831 Remove redundant method, improve docs.
8833 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8835 * docs/libs/gst-rtsp-server-sections.txt:
8836 * gst/rtsp-server/rtsp-auth.h:
8837 * gst/rtsp-server/rtsp-permissions.c:
8838 * gst/rtsp-server/rtsp-permissions.h:
8839 * gst/rtsp-server/rtsp-token.c:
8842 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-permissions.c:
8845 permissions: implement _remove_role
8847 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8849 * gst/rtsp-server/rtsp-permissions.c:
8850 permissions: update docs
8852 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8854 * tests/check/gst/client.c:
8855 tests: simplify tests
8856 Client settings are now disabled by default so we don't need an auth
8857 module to disable them.
8859 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8861 * gst/rtsp-server/rtsp-auth.c:
8862 auth: add default authorizations
8863 When no auth module is specified, use our table of defaults to look up the
8864 default value of the check instead of always allowing everything. This was
8865 we can disallow client settings by default.
8867 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8870 README: update readme
8872 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8874 * gst/rtsp-server/rtsp-thread-pool.c:
8875 * gst/rtsp-server/rtsp-thread-pool.h:
8876 thread-pool: add more docs
8878 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/rtsp-thread-pool.c:
8881 * gst/rtsp-server/rtsp-thread-pool.h:
8882 thread-pool: fix race in thread reuse
8883 If we try to reuse a thread right after we made it stop, we end up using a
8884 stopped thread. Catch this case and only reuse threads that are not stopping.
8886 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8888 * gst/rtsp-server/rtsp-server.c:
8889 server: add small debug
8891 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8893 * tests/check/gst/client.c:
8895 Add some permissions to media so we can use the auth and enable
8898 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8900 * gst/rtsp-server/rtsp-client.c:
8901 client: support pushed context in handle_request
8902 If we already have a pushed state, reuse it and add our own things. This makes
8903 it easier to write tests.
8905 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8907 * gst/rtsp-server/rtsp-auth.c:
8908 auth: don't auth on methods
8909 Don't authorize on methods anymore but on the resources that we
8910 try to access, this is more flexible.
8911 Move the authorization checks to where they are needed and let the
8912 check return the response on error.
8914 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8916 * gst/rtsp-server/rtsp-mount-points.c:
8917 mount-points: add some debug
8919 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8921 * tests/check/gst/client.c:
8922 tests: almost fix test
8924 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8926 * gst/rtsp-server/rtsp-auth.c:
8927 * gst/rtsp-server/rtsp-auth.h:
8928 * gst/rtsp-server/rtsp-client.c:
8929 * gst/rtsp-server/rtsp-client.h:
8930 * gst/rtsp-server/rtsp-server.c:
8931 * gst/rtsp-server/rtsp-server.h:
8932 auth: let the auth module check client_settings
8933 Let the auth module decide if client settings are allowed for the
8936 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8938 * gst/rtsp-server/rtsp-token.c:
8939 * gst/rtsp-server/rtsp-token.h:
8940 token: add method to check boolean permission
8942 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8944 * examples/test-auth.c:
8945 * examples/test-cgroups.c:
8946 * gst/rtsp-server/rtsp-token.c:
8947 * gst/rtsp-server/rtsp-token.h:
8948 token: simplify token constructor
8949 Use variable arguments to make easier API.
8951 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8953 * examples/test-auth.c:
8954 * examples/test-cgroups.c:
8955 * gst/rtsp-server/rtsp-media-factory.c:
8956 * gst/rtsp-server/rtsp-media-factory.h:
8957 media-factory: add convenience API for factory
8959 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8961 * examples/test-auth.c:
8962 * examples/test-cgroups.c:
8963 * gst/rtsp-server/rtsp-permissions.c:
8964 * gst/rtsp-server/rtsp-permissions.h:
8965 permissions: simplify API a little
8966 Avoid passing GstStructure in the add_role method, use varargs instead
8967 to construct the structure behind the scenes. We can then also use the
8968 structure name as the role and simplify some more logic.
8970 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8972 * gst/rtsp-server/rtsp-auth.c:
8975 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8977 * gst/rtsp-server/rtsp-auth.c:
8978 * gst/rtsp-server/rtsp-auth.h:
8979 * gst/rtsp-server/rtsp-client.c:
8980 auth: handle unauthorized response
8981 Move handling of the unauthorized response to the auth module, it can add
8982 the appropriate headers to request authorization for the required method
8983 much better than the client.
8985 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8987 * gst/rtsp-server/rtsp-client.c:
8988 * gst/rtsp-server/rtsp-client.h:
8989 client: allow for sending any message, not only requests
8990 Change the _send_request() method to _send_message() so that we
8991 can both send requests and replies.
8993 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8995 * docs/libs/gst-rtsp-server-sections.txt:
8996 * gst/rtsp-server/rtsp-server.h:
8999 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * examples/test-video.c:
9002 * gst/rtsp-server/rtsp-auth.c:
9003 * gst/rtsp-server/rtsp-auth.h:
9004 * gst/rtsp-server/rtsp-server.c:
9005 * gst/rtsp-server/rtsp-server.h:
9006 auth: move TLS handling to auth module
9007 Remove the TLS settings on the server and move it to the auth module because
9008 that is where security related bits go.
9010 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9012 * gst/rtsp-server/rtsp-client.c:
9013 * gst/rtsp-server/rtsp-client.h:
9014 client: add state push/pop
9016 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9018 * gst/rtsp-server/rtsp-client.c:
9019 * gst/rtsp-server/rtsp-client.h:
9020 client: add connection to state
9022 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9024 * gst/rtsp-server/rtsp-mount-points.c:
9025 mount-points: fix debug
9027 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9029 * tests/check/gst/media.c:
9030 tests: fix media test
9032 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9034 * gst/rtsp-server/rtsp-thread-pool.c:
9035 thread-pool: we don't require a state
9037 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9039 * gst/rtsp-server/rtsp-server.c:
9040 server: let context ref the server
9041 So that we don't risk losing the server object early anc crash.
9043 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9045 * tests/check/gst/client.c:
9046 tests: fix client test
9048 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9051 * docs/libs/gst-rtsp-server-docs.sgml:
9052 * docs/libs/gst-rtsp-server-sections.txt:
9053 * gst/rtsp-server/rtsp-address-pool.c:
9054 * gst/rtsp-server/rtsp-auth.c:
9055 * gst/rtsp-server/rtsp-client.c:
9056 * gst/rtsp-server/rtsp-client.h:
9057 * gst/rtsp-server/rtsp-media-factory-uri.c:
9058 * gst/rtsp-server/rtsp-media-factory.c:
9059 * gst/rtsp-server/rtsp-media-factory.h:
9060 * gst/rtsp-server/rtsp-media.c:
9061 * gst/rtsp-server/rtsp-mount-points.c:
9062 * gst/rtsp-server/rtsp-params.c:
9063 * gst/rtsp-server/rtsp-permissions.c:
9064 * gst/rtsp-server/rtsp-sdp.c:
9065 * gst/rtsp-server/rtsp-server.c:
9066 * gst/rtsp-server/rtsp-server.h:
9067 * gst/rtsp-server/rtsp-session-media.c:
9068 * gst/rtsp-server/rtsp-session-pool.c:
9069 * gst/rtsp-server/rtsp-session.c:
9070 * gst/rtsp-server/rtsp-stream-transport.c:
9071 * gst/rtsp-server/rtsp-stream.c:
9072 * gst/rtsp-server/rtsp-thread-pool.c:
9073 * gst/rtsp-server/rtsp-token.c:
9076 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9078 * gst/rtsp-server/rtsp-session-pool.c:
9079 * gst/rtsp-server/rtsp-session-pool.h:
9080 session-pool: make vmethod to create a session
9081 Make a vmethod to create a sessions so that subclasses can create
9082 custom session objects
9084 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9086 * gst/rtsp-server/rtsp-auth.c:
9087 * gst/rtsp-server/rtsp-media-factory.h:
9088 * gst/rtsp-server/rtsp-media.h:
9089 * gst/rtsp-server/rtsp-mount-points.h:
9090 * gst/rtsp-server/rtsp-session-pool.h:
9091 * gst/rtsp-server/rtsp-stream.h:
9094 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9096 * docs/libs/gst-rtsp-server-docs.sgml:
9097 * docs/libs/gst-rtsp-server-sections.txt:
9098 * gst/rtsp-server/rtsp-address-pool.c:
9099 * gst/rtsp-server/rtsp-address-pool.h:
9100 * gst/rtsp-server/rtsp-auth.c:
9101 * gst/rtsp-server/rtsp-client.h:
9102 * gst/rtsp-server/rtsp-media-factory.h:
9103 * gst/rtsp-server/rtsp-media.c:
9104 * gst/rtsp-server/rtsp-media.h:
9105 * gst/rtsp-server/rtsp-permissions.c:
9106 * gst/rtsp-server/rtsp-permissions.h:
9107 * gst/rtsp-server/rtsp-server.h:
9108 * gst/rtsp-server/rtsp-session-media.c:
9109 * gst/rtsp-server/rtsp-session-media.h:
9110 * gst/rtsp-server/rtsp-session-pool.h:
9111 * gst/rtsp-server/rtsp-session.h:
9112 * gst/rtsp-server/rtsp-stream-transport.h:
9113 * gst/rtsp-server/rtsp-stream.c:
9114 * gst/rtsp-server/rtsp-thread-pool.h:
9117 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9120 * examples/Makefile.am:
9121 configure: compile cgroup example conditionally
9122 Only compile the cgroup example when we have libcgroup
9124 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9127 * examples/Makefile.am:
9128 * examples/test-cgroups.c:
9129 examples: add cgroups example
9131 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9133 * tests/check/gst/rtspserver.c:
9134 tests: fix compilation
9136 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9138 * gst/rtsp-server/rtsp-thread-pool.c:
9139 thread-pool: fix vmethod invocation
9141 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9143 * gst/rtsp-server/rtsp-thread-pool.c:
9144 * gst/rtsp-server/rtsp-thread-pool.h:
9145 thread-pool: store thread type in thread
9147 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9149 * gst/rtsp-server/rtsp-client.c:
9150 client: pass thread from pool to media _prepare
9151 Get a thread from the configured threadpool and pass it to the prepare method of
9154 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9156 * gst/rtsp-server/rtsp-media.c:
9157 * gst/rtsp-server/rtsp-media.h:
9158 media: Accept a thread in _prepare
9159 Remove out own threadpool handling and use the provided thread and
9160 maincontext for the bus messages and the state changes.
9162 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9164 * gst/rtsp-server/rtsp-server.c:
9165 server: configure client thread pool
9167 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9169 * gst/rtsp-server/rtsp-client.c:
9170 * gst/rtsp-server/rtsp-client.h:
9171 client: add method to configure thread pool
9173 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9175 * gst/rtsp-server/rtsp-client.h:
9176 * gst/rtsp-server/rtsp-server.c:
9177 * gst/rtsp-server/rtsp-server.h:
9178 server: use thread pool
9179 Use the thread pool instead of doing our own thing.
9181 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9183 * gst/rtsp-server/Makefile.am:
9184 * gst/rtsp-server/rtsp-thread-pool.c:
9185 * gst/rtsp-server/rtsp-thread-pool.h:
9186 thread-pool: add object to manage threads
9187 Add an object to manage the client and media threads.
9189 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9191 * gst/rtsp-server/rtsp-auth.c:
9192 auth: debug authorization check
9194 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9196 * gst/rtsp-server/rtsp-media.c:
9197 media: start media pipeline in context
9198 Start the media pipeline in the provided context (or our default one
9199 when NULL). This makes sure that we run the bus thread in this context and that
9200 all media threads are children of this context.
9202 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9204 * gst/rtsp-server/rtsp-media-factory.c:
9205 factory: pass permissions to media by default
9207 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9209 * examples/test-auth.c:
9210 test: add permissions to auth test
9211 Ass some permissions to the media factory in the test.
9213 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9215 * gst/rtsp-server/rtsp-auth.c:
9216 * gst/rtsp-server/rtsp-auth.h:
9217 * gst/rtsp-server/rtsp-client.c:
9218 auth: simplify auth checks
9219 Remove client from methods, it's now in the state
9220 Perform the check specified by the string, use the information from the
9221 thread local context.
9223 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9225 * gst/rtsp-server/rtsp-client.c:
9226 * gst/rtsp-server/rtsp-client.h:
9227 client: add state to current thread
9228 Add the client to the ClientState object.
9229 Place the ClientState on the current thread.
9231 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9233 * gst/rtsp-server/rtsp-media-factory.c:
9234 * gst/rtsp-server/rtsp-media-factory.h:
9235 * gst/rtsp-server/rtsp-media.c:
9236 * gst/rtsp-server/rtsp-media.h:
9237 media: make it possible to set permissions
9238 Make it possible to set permissions on media and media factory objects
9240 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9242 * gst/rtsp-server/Makefile.am:
9243 * gst/rtsp-server/rtsp-permissions.c:
9244 * gst/rtsp-server/rtsp-permissions.h:
9245 permissions: add permissions object
9246 Add a mini object to store permissions based on a role.
9248 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9250 * examples/test-auth.c:
9251 * gst/rtsp-server/rtsp-auth.c:
9252 * gst/rtsp-server/rtsp-auth.h:
9253 * gst/rtsp-server/rtsp-client.c:
9254 auth: add auth checks
9255 Add an enum with auth checks and implement the checks in the auth object.
9256 Perform the checks from the client.
9258 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9260 * examples/test-auth.c:
9261 * gst/rtsp-server/rtsp-auth.c:
9262 * gst/rtsp-server/rtsp-auth.h:
9263 * gst/rtsp-server/rtsp-client.h:
9264 auth: use the token after authentication
9265 After we authenticated a user, keep the Token around in the state.
9267 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9269 * gst/rtsp-server/rtsp-client.c:
9270 * gst/rtsp-server/rtsp-media.c:
9271 * gst/rtsp-server/rtsp-media.h:
9272 * tests/check/gst/media.c:
9273 media: add optional context for bus messages
9274 Add an optional mainloop to _prepare that will handle the bus messages instead
9275 of always using the shared mainloop.
9277 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9279 * gst/rtsp-server/Makefile.am:
9280 * gst/rtsp-server/rtsp-token.c:
9281 * gst/rtsp-server/rtsp-token.h:
9282 token: add authorization token
9283 Add a simply miniobject that contains the authorizations. The object contains a
9284 GstStructure that hold all authorization fields. When a user is authenticated,
9285 the auth module will create a Token for the user. The token is then used to
9286 check what operations the user is allowed to do and various other configuration
9289 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9291 * examples/test-auth.c:
9292 * gst/rtsp-server/rtsp-auth.c:
9293 * gst/rtsp-server/rtsp-auth.h:
9294 * gst/rtsp-server/rtsp-client.c:
9295 * gst/rtsp-server/rtsp-client.h:
9296 * gst/rtsp-server/rtsp-media-factory.c:
9297 * gst/rtsp-server/rtsp-media-factory.h:
9298 * gst/rtsp-server/rtsp-media.c:
9299 * gst/rtsp-server/rtsp-media.h:
9300 auth: remove auth from media and factory
9301 Remove the auth object from media and factory. We want to have the RTSPClient
9302 authenticate and authorize resources, there is no need to place another auth
9303 manager on the media/factory.
9305 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9307 * examples/test-auth.c:
9308 * gst/rtsp-server/rtsp-auth.c:
9309 * gst/rtsp-server/rtsp-auth.h:
9310 * gst/rtsp-server/rtsp-client.h:
9311 auth: add support for multiple basic auth tokens
9312 Make it possible to add multiple basic authorisation tokens to one authorization
9313 object. Associate with each token an authorization group that will define what
9314 capabilities are allowed.
9316 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9318 * gst/rtsp-server/rtsp-client.c:
9319 client: error out on non-aggregate control
9320 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9322 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9324 * gst/rtsp-server/rtsp-client.c:
9325 client: rework setup request a little
9326 Cache the media in DESCRIBE based on the longest matching path with the uri
9327 that we can find in the mount points.
9328 Rework the setup request a little to get the media from the session or from
9329 the longest matching path, this way we can derive the control string as
9330 everything after the path instead of hardcoding it.
9331 Find the stream based on the control string and only open a session when all
9334 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9336 * gst/rtsp-server/rtsp-media.c:
9337 * gst/rtsp-server/rtsp-media.h:
9338 media: add method to find a stream by control url
9340 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9342 * gst/rtsp-server/rtsp-stream.c:
9343 * gst/rtsp-server/rtsp-stream.h:
9344 stream: add method to check control url of stream
9346 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9348 * gst/rtsp-server/rtsp-client.c:
9349 * gst/rtsp-server/rtsp-session-media.c:
9350 * gst/rtsp-server/rtsp-session-media.h:
9351 * gst/rtsp-server/rtsp-session.c:
9352 * gst/rtsp-server/rtsp-session.h:
9353 session: use path matching for session media
9354 Use a path string instead of a uri to lookup session media in the sessions. Also
9355 use path matching to find the largest possible path that matches.
9357 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9359 * gst/rtsp-server/rtsp-client.c:
9360 * gst/rtsp-server/rtsp-mount-points.c:
9361 * gst/rtsp-server/rtsp-mount-points.h:
9362 * tests/check/gst/mountpoints.c:
9363 mount-points: remove useless vmethod
9364 Making lookups in the mount points should not be done with a URL, if there is a
9365 mapping to be done from URL to mount points, we'll need to do it somewhere
9368 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9370 * gst/rtsp-server/rtsp-mount-points.c:
9371 * gst/rtsp-server/rtsp-mount-points.h:
9372 * tests/check/gst/mountpoints.c:
9373 mount-points: improve mount point searching
9374 Use a GSequence to keep track of the mount points.
9375 Match a URL to the longest matching registered mount point. This should be the
9376 URL to perform aggreagate control and the remainder is the stream specific
9378 Add some unit tests for this.
9380 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9382 * gst/rtsp-server/Makefile.am:
9383 rtsp-server: Allow building of static library
9385 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9387 * tests/check/gst/mediafactory.c:
9388 tests: fix compilation
9390 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9392 * gst/rtsp-server/rtsp-sdp.c:
9393 sdp: get control string from stream
9394 Use the control string as configured in the stream.
9396 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9398 * gst/rtsp-server/rtsp-stream.c:
9399 * gst/rtsp-server/rtsp-stream.h:
9400 stream: add methods and property to set control string
9402 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9404 * gst/rtsp-server/rtsp-client.c:
9406 Rename variables for clarity
9407 Keep media in state when we can
9409 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * gst/rtsp-server/rtsp-client.c:
9412 * gst/rtsp-server/rtsp-stream.c:
9413 * gst/rtsp-server/rtsp-stream.h:
9414 stream: add more support for IPv6
9415 Rename _get_address to _get_multicast_address in GstRTSPStream to
9416 make it clear that this function only deals with multicast.
9417 Make it possible to have both an IPv4 and IPv6 multicast address on
9418 a stream. Give the client an IPv4 or IPv6 address depending on the
9419 address it used to connect to the server.
9420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9422 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9424 * gst/rtsp-server/rtsp-client.c:
9427 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9429 * gst/rtsp-server/rtsp-stream.c:
9430 stream: handle failed port allocation
9431 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9432 can't allocate any family at all. Also keep track of what port families we
9434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9436 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9438 * gst/rtsp-server/rtsp-stream.c:
9439 stream: improve docs
9441 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9443 * gst/rtsp-server/rtsp-stream-transport.c:
9444 stream-transport: remove old if 0 block
9446 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9448 * tests/check/gst/client.c:
9450 gst_rtsp_client_get_uri() has been removed
9451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9453 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9455 * gst/rtsp-server/rtsp-client.c:
9456 * gst/rtsp-server/rtsp-client.h:
9457 client: add method to filter managed sessions
9458 Add a method to filter the sessions managed by this client connection.
9459 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9461 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9463 * gst/rtsp-server/rtsp-client.c:
9464 * gst/rtsp-server/rtsp-client.h:
9465 client: remove _get_uri() method
9466 Remove the get_uri() method on the client. A client has no uri, the uri
9467 property is an internal property to manage the last cached media for
9470 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9472 * gst/rtsp-server/rtsp-media-factory.h:
9473 media-factory: fix typo
9475 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9477 * gst/rtsp-server/rtsp-media.c:
9478 rtsp-media: Do not leak the query in default_query_stop
9479 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9481 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9483 * gst/rtsp-server/rtsp-media.c:
9484 media: don't unlock when conversion fails
9485 Don't unlock the state lock when conversion fails because it was not locked.
9487 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9489 * gst/rtsp-server/rtsp-media.c:
9490 * gst/rtsp-server/rtsp-media.h:
9491 Add query_position and query_stop vmethods to rtsp-media
9493 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9495 * gst/rtsp-server/rtsp-media.c:
9496 Fix typo in property install for rtsp-media's time-provider
9498 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9500 * gst/rtsp-server/rtsp-client.c:
9501 * gst/rtsp-server/rtsp-client.h:
9502 client: clean some variables
9503 Clean some variables and add some guards to _send_request()
9505 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9507 * gst/rtsp-server/rtsp-client.c:
9508 * gst/rtsp-server/rtsp-client.h:
9509 Add gst_rtsp_client_send_request API
9510 This makes it possible to send arbitrary messages to a client, such as
9511 SET_PARAMETER or GET_PARAMETER
9513 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9515 * gst/rtsp-server/rtsp-media.c:
9516 * gst/rtsp-server/rtsp-media.h:
9517 media: add _get_element() method
9518 Add method to get the element used when creating the media.
9519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9521 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9523 * gst/rtsp-server/rtsp-media.c:
9526 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9528 * gst/rtsp-server/rtsp-stream.c:
9529 * gst/rtsp-server/rtsp-stream.h:
9530 stream: allow access to the rtp session
9531 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9533 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9535 * gst/rtsp-server/rtsp-stream.c:
9536 * gst/rtsp-server/rtsp-stream.h:
9537 dscp qos support in gst-rtsp-stream
9538 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9540 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9542 * tests/check/gst/rtspserver.c:
9544 Actually do what the comment says. Also keep the old code around, not sure what
9545 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9546 it currently doesn't.
9548 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9550 * gst/rtsp-server/rtsp-client.c:
9551 client: also watch newly created session
9552 When we newly created a session, start watching it immediately instead of
9553 on the next request.
9555 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9557 * tests/check/gst/client.c:
9558 tests: add unit test for new-session
9559 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9561 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9563 * gst/rtsp-server/rtsp-client.c:
9564 client: emit new-session when new session is created
9565 Only emit new-session when we created a new session for a client, not when a
9566 client picked up a previous session.
9567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9569 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9571 * gst/rtsp-server/rtsp-client.c:
9572 client: handle asterisk as path in requests
9573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9575 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9577 * gst/rtsp-server/rtsp-media.c:
9578 media: handle segment query format mismatch
9579 It's possible that the segment query returns with a different format than what
9580 we asked for, handle this case also.
9582 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9584 * gst/rtsp-server/rtsp-media.c:
9585 media: use segment stop in collect_media_stats
9586 Use segment stop instead of duration as range end point.
9587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9589 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9591 * gst/rtsp-server/rtsp-media.c:
9592 * tests/check/gst/media.c:
9593 rtsp-media: Do not leak the element in take_pipeline
9594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9596 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9598 * gst/rtsp-server/rtsp-client.c:
9599 * gst/rtsp-server/rtsp-client.h:
9600 rtsp-client: Make configure_client_transport virtual
9601 This patch makes configure_client_transport virtual. The functionality is
9602 needed to handle some weird clients sending multicast transport settings as url
9604 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9606 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9608 * gst/rtsp-server/rtsp-client.c:
9609 * gst/rtsp-server/rtsp-client.h:
9610 rtsp-client: Make param_set and param_get virtual
9611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9613 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9615 * gst/rtsp-server/rtsp-client.c:
9616 * gst/rtsp-server/rtsp-media.c:
9617 * gst/rtsp-server/rtsp-media.h:
9618 media: convert_range replaces get_range_times
9619 get_range_times worked for handling UTC ranges for seeks, but we also
9620 need to convert back from NPT to the requested unit in
9621 get_range_string. convert_range is now used for both.
9622 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9624 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9626 * gst/rtsp-server/rtsp-client.c:
9627 * gst/rtsp-server/rtsp-sdp.c:
9628 * gst/rtsp-server/rtsp-sdp.h:
9629 sdp: cleanup sdp info
9630 We don't need to pass the proto, we can more easily check a boolean.
9631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9633 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9635 * gst/rtsp-server/rtsp-sdp.c:
9636 use 0.0.0.0 or :: for c= line instead of server address
9638 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9640 * gst/rtsp-server/rtsp-client.c:
9641 use local address, not remote, in SDP
9642 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9644 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9647 Automatic update of common submodule
9648 From 098c0d7 to 01a7a46
9650 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9652 * gst/rtsp-server/rtsp-media.c:
9653 * gst/rtsp-server/rtsp-media.h:
9654 media: possibility to override range time conversion
9655 Make it possible to override the conversion from GstRTSPTimeRange to
9656 GstClockTimes, that is done before seeking on the media
9657 pipeline. Overriding can be useful for UTC ranges, where the default
9658 conversion gives nanoseconds since 1900.
9659 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9661 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9663 * gst/rtsp-server/rtsp-server.c:
9664 * gst/rtsp-server/rtsp-server.h:
9665 rtsp-server: Expose the use_client_settings API
9666 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9668 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9670 * gst/rtsp-server/rtsp-client.c:
9671 * gst/rtsp-server/rtsp-stream.c:
9672 * gst/rtsp-server/rtsp-stream.h:
9673 rtspstream: handle both ipv4 and ipv6 clients
9674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9676 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9678 * gst/rtsp-server/rtsp-sdp.c:
9679 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9680 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9681 We already have a way to place extra attributes in the SDP by using a string
9682 property with prefix x- or a- in the caps.
9684 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9686 * gst/rtsp-server/rtsp-sdp.c:
9687 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9688 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9689 We already have a way to place extra attributes in the SDP, just make a string
9690 property in the payloader with a- or x- prefix.
9692 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9694 * gst/rtsp-server/rtsp-sdp.c:
9695 rtsp: place a- and x- properties as attributes
9696 application/x-rtp has properties with a- and x- prefixes that should be
9697 placed as attributes in the SDP for the media instead of being added to the
9700 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9702 * examples/Makefile.am:
9703 * examples/test-video.c:
9704 example: add TLS example
9706 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9708 * gst/rtsp-server/rtsp-server.c:
9709 * gst/rtsp-server/rtsp-server.h:
9710 server: add support for TLS
9711 Add methods to set and get a TLS certificate.
9712 Add vmethod to configure a new connection. By default, configure the TLS
9713 certificate in a new connection if needed.
9715 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9717 * gst/rtsp-server/rtsp-server.c:
9718 * gst/rtsp-server/rtsp-server.h:
9719 server: remove accept_client vmethod
9720 This vmethod is not very useful so remove it.
9722 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9724 * gst/rtsp-server/rtsp-server.c:
9725 server: don't crash on NULL GError
9727 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
9729 * gst/rtsp-server/rtsp-session-pool.c:
9730 rtsp-session-pool: corrected session timeout detection
9731 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
9733 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9735 * gst/rtsp-server/rtsp-client.c:
9736 client: improve debug
9738 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9740 * gst/rtsp-server/rtsp-client.c:
9741 * gst/rtsp-server/rtsp-client.h:
9742 * gst/rtsp-server/rtsp-server.c:
9743 server: refactor connection setup
9744 Let the server accept the socket connection and construct a GstRTSPConnection
9745 from it. Remove the code from the client and let the client only deal with
9746 a fully configure GstRTSPConnection object.
9747 We will need this later when the server will configure the connection for
9750 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9752 * gst/rtsp-server/rtsp-stream.c:
9753 stream: keep the transport object alive
9754 Keep the transport object alive while we have it as qdata on the
9757 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
9759 * gst/rtsp-server/rtsp-client.c:
9760 * gst/rtsp-server/rtsp-server.c:
9761 rtsp-server: Do not crash on nmapping of server
9762 * generate error when gst_rtsp_connection_accept fails
9763 * do not stop accepting incoming connections because
9764 accepting a client fails
9765 https://bugzilla.gnome.org/show_bug.cgi?id=701072
9767 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
9769 * gst/rtsp-server/rtsp-client.c:
9770 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
9771 https://bugzilla.gnome.org/show_bug.cgi?id=700953
9773 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9775 * gst/rtsp-server/rtsp-sdp.c:
9776 rtsp-sdp: Parse framerate caps field and set SDP attribute
9777 The SDP attribute and its format is described in RFC4566.
9778 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9780 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
9782 * gst/rtsp-server/rtsp-sdp.c:
9783 rtsp-sdp: Parse width/height from caps and set SDP attribute
9784 The SDP attribute and its format is described in RFC6064.
9785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9787 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
9789 * gst/rtsp-server/rtsp-sdp.c:
9790 * tests/check/gst/client.c:
9791 rtsp-sdp: add bandwidth line
9792 https://bugzilla.gnome.org/show_bug.cgi?id=699220
9794 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9797 Automatic update of common submodule
9798 From 5edcd85 to 098c0d7
9800 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9802 * tests/check/gst/media.c:
9803 tests: add dynamic payloader prepare/unprepare check
9805 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9807 * gst/rtsp-server/rtsp-media.c:
9808 media: release lock when removing fakesink
9810 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9812 * gst/rtsp-server/rtsp-stream.c:
9813 stream: set elements to NULL before removing
9814 When removing a stream, set the elements to NULL first. This avoids
9815 element-is-not-in-NULL-state errors when we dispose the elements.
9817 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9820 Automatic update of common submodule
9821 From 3cb3d3c to 5edcd85
9823 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9825 * gst/rtsp-server/rtsp-media.c:
9826 * gst/rtsp-server/rtsp-media.h:
9827 media: listen to pad-removed signals
9828 Listen to the pad-removed signal and remove the stream associated with the
9830 Add signal to be notified of the removed pad.
9831 Remove the fakesink in unprepare()
9832 Fix signatures of the signal methods
9834 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9836 * examples/test-sdp.c:
9837 tests: add example of reusable pipelines
9839 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
9841 * gst/rtsp-server/rtsp-stream.c:
9842 * gst/rtsp-server/rtsp-stream.h:
9843 stream: add method to get the srcpad
9845 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9847 * tests/check/gst/media.c:
9848 check: add media prepare/unprepare test
9849 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9851 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
9853 * gst/rtsp-server/rtsp-media.c:
9854 media: disconnect from signal handlers in unprepare()
9855 We connected to the pad-added and no-more-pads signals in prepare() so
9856 we need to disconnect from them in unprepare().
9857 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9859 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
9861 * gst/rtsp-server/rtsp-media.c:
9862 media: don't free streams array
9863 Don't free the streams array in the unprepare() method, they were not
9865 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9867 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
9869 * gst/rtsp-server/rtsp-media.c:
9870 media: don't unref the pipeline in unprepare
9871 Unprepare() should undo what prepare() does. Because the pipeline is
9872 not created in prepare(), we should not unref it in unprepare()
9874 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
9876 * gst/rtsp-server/rtsp-stream.c:
9877 stream: clear session and caps for reuse
9878 Set the session and caps to NULL after unref otherwise we might unref
9880 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9882 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
9884 * gst/rtsp-server/rtsp-client.c:
9885 client: send out teardown signal before tearing down
9886 The advantage is that in the signal handler you get direct access to
9887 information about what streams are about to get torn down (in the
9888 GstRTSPClientState).
9889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
9891 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
9893 * gst/rtsp-server/rtsp-client.c:
9894 * gst/rtsp-server/rtsp-client.h:
9895 client: expose connection
9896 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
9898 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
9901 Automatic update of common submodule
9902 From aed87ae to 3cb3d3c
9904 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9906 * gst/rtsp-server/rtsp-media.c:
9907 * gst/rtsp-server/rtsp-media.h:
9908 * gst/rtsp-server/rtsp-session-media.c:
9909 * gst/rtsp-server/rtsp-session-media.h:
9910 media: add method to get the base_time of the pipeline
9911 Together with a shared clock, this base-time could eventually be sent to
9912 the client so that it can reconstruct the exact running-time of the clock
9915 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9917 * gst/rtsp-server/Makefile.am:
9918 * gst/rtsp-server/rtsp-media.c:
9919 * gst/rtsp-server/rtsp-media.h:
9920 * gst/rtsp-server/rtsp-sdp.c:
9921 media: add GstNetTimeProvider support
9922 Add a property to let the media provide a GstNetTimeProvider for its clock.
9923 Make methods to get the clock and nettimeprovider
9924 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
9925 provider and also the current time of the clock. This should make it possible
9926 for (GStreamer) clients to slave their clock to the server clock.
9928 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
9931 Automatic update of common submodule
9932 From 04c7a1e to aed87ae
9934 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9936 * gst/rtsp-server/rtsp-media.c:
9937 media: wait for buffering to complete
9938 Wait for buffering to complete before changing the state to the target state.
9940 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9942 * gst/rtsp-server/rtsp-media.c:
9943 media: small cleanup
9945 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
9947 * tests/check/gst/rtspserver.c:
9948 tests: remove extra unref in test_setup_non_existing_stream
9949 The unref is not needed anymore, teardown runs without it.
9950 https://bugzilla.gnome.org/show_bug.cgi?id=696542
9952 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
9954 * tests/check/gst/rtspserver.c:
9955 tests: GSocketService cleanup in test_bind_already_in_use
9956 Use g_socket_service_stop so the rtspserver test stops listening for
9957 incoming connections in test_bind_already_in_use.
9958 https://bugzilla.gnome.org/show_bug.cgi?id=696541
9960 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
9962 * gst/rtsp-server/rtsp-media-factory.c:
9963 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
9964 Instead use a GWeakRef which is safe to use
9965 This is a known GLib bug, see:
9966 https://bugzilla.gnome.org/show_bug.cgi?id=667145
9968 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
9970 * gst/rtsp-server/rtsp-client.c:
9971 * gst/rtsp-server/rtsp-media.c:
9972 * gst/rtsp-server/rtsp-media.h:
9973 * gst/rtsp-server/rtsp-sdp.c:
9974 * tests/check/gst/media.c:
9975 * tests/check/gst/rtspserver.c:
9976 rtsp-media/client: Reply to PLAY request with same type of Range
9977 Remember the type of Range from the PLAY request and use the same type for
9980 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
9982 * gst/rtsp-server/rtsp-client.c:
9983 * gst/rtsp-server/rtsp-client.h:
9984 * tests/check/gst/client.c:
9985 rtsp-client: expose uri
9987 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
9989 * tests/check/gst/mediafactory.c:
9990 tests: Hold ref while creating second media
9991 To test if the media aren't shared, make sure we keep the first one while creating a second
9992 otherwise the same memory address may be reused.
9994 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
9997 configure: remove out-of-date comment
9999 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10002 .gitignore: ignore more build files
10004 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10006 * tests/check/Makefile.am:
10007 tests: use right _LIBS variable for gst-plugins-base libs
10009 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10011 * tests/check/Makefile.am:
10012 check: add librtp to libs
10014 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10016 * tests/check/gst/rtspserver.c:
10017 tests: Add test to check selecting a port the server will send from
10019 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10021 * tests/check/gst/rtspserver.c:
10022 tests: Make sure packets are actually received
10024 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10026 * gst/rtsp-server/rtsp-stream.c:
10027 stream: Select unicast address from pool if appropriate
10029 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10031 * gst/rtsp-server/rtsp-stream.c:
10032 stream: Properties are always there in Gst 1.0
10034 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10036 * tests/check/gst/addresspool.c:
10037 tests: Add tests for unicast addresses in pool
10039 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10041 * gst/rtsp-server/rtsp-address-pool.c:
10042 * tests/check/gst/addresspool.c:
10043 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10045 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10047 * docs/libs/gst-rtsp-server-sections.txt:
10048 * gst/rtsp-server/rtsp-address-pool.c:
10049 * gst/rtsp-server/rtsp-address-pool.h:
10050 * gst/rtsp-server/rtsp-stream.c:
10051 * tests/check/gst/addresspool.c:
10052 address-pool: Add unicast addresses
10054 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10057 * gst/rtsp-server/rtsp-server.c:
10058 * tests/check/gst/rtspserver.c:
10059 rtsp-server: Limit the number of threads per server instance
10060 If we exceed the maximum, just round robin the clients over the existing
10063 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10065 * gst/rtsp-server/rtsp-server.c:
10066 rtsp-server: No need to store the GMainContext in the client context
10068 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10070 * tests/check/gst/rtspserver.c:
10071 tests: Add test for client disconnection
10073 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10075 * tests/check/gst/rtspserver.c:
10076 tests: Test client and session timeouts with multiple threads
10078 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10080 * gst/rtsp-server/rtsp-address-pool.c:
10081 * gst/rtsp-server/rtsp-auth.c:
10082 * gst/rtsp-server/rtsp-client.c:
10083 * gst/rtsp-server/rtsp-media-factory-uri.c:
10084 * gst/rtsp-server/rtsp-media-factory.c:
10085 * gst/rtsp-server/rtsp-media.c:
10086 * gst/rtsp-server/rtsp-mount-points.c:
10087 * gst/rtsp-server/rtsp-server.c:
10088 * gst/rtsp-server/rtsp-session-media.c:
10089 * gst/rtsp-server/rtsp-session-pool.c:
10090 * gst/rtsp-server/rtsp-session.c:
10091 Document locking and its order
10093 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10095 * tests/check/gst/rtspserver.c:
10096 tests: Test that slow DESCRIBE don't block other clients
10098 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10100 * tests/check/gst/client.c:
10101 tests: Add tests for client-requested multicast address
10103 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10105 * docs/libs/gst-rtsp-server-sections.txt:
10106 docs: Put the various functions in the right sections
10108 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10110 * docs/libs/gst-rtsp-server-docs.sgml:
10111 * docs/libs/gst-rtsp-server-sections.txt:
10112 * gst/rtsp-server/rtsp-address-pool.c:
10113 * gst/rtsp-server/rtsp-address-pool.h:
10114 docs: Generate docs for GstRTSPAddressPool
10116 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10118 * gst/rtsp-server/rtsp-client.c:
10119 * gst/rtsp-server/rtsp-stream.c:
10120 * gst/rtsp-server/rtsp-stream.h:
10121 client: Check client provided addresses against the address pool
10123 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10125 * gst/rtsp-server/rtsp-address-pool.c:
10126 * gst/rtsp-server/rtsp-address-pool.h:
10127 * tests/check/gst/addresspool.c:
10128 address-pool: Add API to request a specific address from the pool
10129 Also add relevant unit tests.
10131 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10133 * tests/check/gst/mediafactory.c:
10134 tests: Check the passing around of a RTSPAddressPool
10135 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10136 way down to the stream.
10138 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10140 * tests/check/gst/addresspool.c:
10141 tests: Add more tests for the address pool
10143 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10145 * gst/rtsp-server/rtsp-address-pool.c:
10146 address-pool: Fix off by one error
10147 When splitting a port range, the port after a skip is not part of range.
10149 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10152 Automatic update of common submodule
10153 From 2de221c to 04c7a1e
10155 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10158 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10159 AM_CONFIG_HEADER was removed in automake 1.13
10160 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10162 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10165 Automatic update of common submodule
10166 From a942293 to 2de221c
10168 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10170 * gst/rtsp-server/rtsp-client.c:
10171 client: make sure the watch exists while sending data
10172 Protect the send_func with a lock. This allows us to wait for sending
10173 to complete before changing the send_func and user_data. We add an
10174 extra ref to the watch to make sure that it remains valid during
10176 When closing the connection, set the send_func to NULL
10177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10179 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10181 * tests/check/Makefile.am:
10182 tests: use GST_*_1_0 environment variables everywhere
10183 The _1_0 suffixed environment variables override the
10184 non-suffixed ones, so if we're in an environment that
10185 sets the _1_0 suffixed ones, such as jhbuild, we need
10186 to set those to make sure ours actually always get
10189 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10192 Automatic update of common submodule
10193 From acb04d9 to a942293
10195 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10197 * gst/rtsp-server/rtsp-client.c:
10198 rtsp-client: set the client backlog
10199 Set the client backlog to a reasonable default
10201 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10203 * gst/rtsp-server/rtsp-media.c:
10204 rtsp-media: Make the element a constructor parameter
10205 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10207 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10209 * docs/libs/Makefile.am:
10210 docs: Link with gcov library when gcov is enabled
10211 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10213 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10215 * gst/rtsp-server/rtsp-media.c:
10216 media: match prepare with unprepare
10217 Really unprepare when there were an equal amount of prepare calls.
10219 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10221 * gst/rtsp-server/rtsp-media.c:
10222 media: media has to be unprepared in finalize
10223 Because unprepare takes away the last ref on the media.
10225 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10227 * gst/rtsp-server/rtsp-client.c:
10228 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10229 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10230 We can't use the refcount to trigger unprepare because it is the unprepare call
10231 that removes the last refcount after all messages are consumed. What we should
10232 probably do is make a prepared refcount and only unprepare when the refcount
10235 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10237 * gst/rtsp-server/rtsp-media.c:
10238 media: let the source unref the last media ref
10239 the last ref to the media is held by the source so we don't need to add more ref
10240 and unrefs, we simply destroy the media when the source is gone.
10242 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10244 * gst/rtsp-server/rtsp-media.c:
10245 media: improve debug
10247 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10249 * gst/rtsp-server/rtsp-media.c:
10251 Make sure we are in the right state when collecting the position and duration.
10252 Only make ourselves PREPARED when we were previously PREPARING.
10254 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10256 * gst/rtsp-server/rtsp-media.c:
10257 media: use g_object_ref/unref for GObjects
10259 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10261 * gst/rtsp-server/rtsp-client.c:
10262 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10263 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10264 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10265 isn't being used anymore.
10267 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10269 * gst/rtsp-server/rtsp-media.c:
10270 Fix compiler warning
10272 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10274 * gst/rtsp-server/rtsp-media-factory-uri.c:
10275 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10277 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10279 * gst/rtsp-server/rtsp-session-media.h:
10282 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10284 * gst/rtsp-server/rtsp-media.c:
10285 * tests/check/gst/media.c:
10286 media: avoid element leak
10288 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10290 * gst/rtsp-server/rtsp-media.c:
10291 media: require an element in media constructor
10293 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10295 * gst/rtsp-server/rtsp-client.c:
10296 Revert "client: TEARDOWN brings that state to Init again"
10297 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10298 The object is already disposed, there is no point in setting the state.
10300 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10302 * gst/rtsp-server/rtsp-client.c:
10303 client: TEARDOWN brings that state to Init again
10305 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10307 * docs/libs/gst-rtsp-server-sections.txt:
10308 * examples/test-auth.c:
10309 * gst/rtsp-server/rtsp-auth.c:
10310 * gst/rtsp-server/rtsp-auth.h:
10311 * gst/rtsp-server/rtsp-client.c:
10312 * gst/rtsp-server/rtsp-client.h:
10313 * gst/rtsp-server/rtsp-media-factory-uri.c:
10314 * gst/rtsp-server/rtsp-media-factory-uri.h:
10315 * gst/rtsp-server/rtsp-media-factory.c:
10316 * gst/rtsp-server/rtsp-media-factory.h:
10317 * gst/rtsp-server/rtsp-media.c:
10318 * gst/rtsp-server/rtsp-media.h:
10319 * gst/rtsp-server/rtsp-mount-points.c:
10320 * gst/rtsp-server/rtsp-mount-points.h:
10321 * gst/rtsp-server/rtsp-sdp.c:
10322 * gst/rtsp-server/rtsp-server.c:
10323 * gst/rtsp-server/rtsp-server.h:
10324 * gst/rtsp-server/rtsp-session-media.c:
10325 * gst/rtsp-server/rtsp-session-media.h:
10326 * gst/rtsp-server/rtsp-session-pool.c:
10327 * gst/rtsp-server/rtsp-session-pool.h:
10328 * gst/rtsp-server/rtsp-session.c:
10329 * gst/rtsp-server/rtsp-session.h:
10330 * gst/rtsp-server/rtsp-stream-transport.c:
10331 * gst/rtsp-server/rtsp-stream-transport.h:
10332 * gst/rtsp-server/rtsp-stream.c:
10333 * gst/rtsp-server/rtsp-stream.h:
10334 * tests/check/gst/media.c:
10335 rtsp: make object details private
10336 Make all object details private
10337 Add methods to access private bits
10339 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10341 * tests/check/Makefile.am:
10342 * tests/check/gst/media.c:
10343 tests: add media tests
10345 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10347 * gst/rtsp-server/rtsp-media.c:
10348 media: check if prepared for some methods
10349 Check that the media object is prepared before doing seek and getting the
10350 current position etc.
10351 Add some g_return checks.
10353 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10355 * tests/check/Makefile.am:
10356 * tests/check/gst/mediafactory.c:
10357 tests: add mediafactory test
10359 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10361 * gst/rtsp-server/rtsp-stream.c:
10362 stream: improve debug
10364 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10366 * gst/rtsp-server/rtsp-media.c:
10367 * gst/rtsp-server/rtsp-media.h:
10368 media: unref pipeline in finalize to avoid leaking it
10370 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10372 * gst/rtsp-server/rtsp-media-factory-uri.c:
10373 * gst/rtsp-server/rtsp-media.c:
10374 rtsp: use gst_object_unref on GstObjects
10376 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10378 * gst/rtsp-server/rtsp-media-factory.c:
10379 media-factory: require an url
10381 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10383 * examples/test-uri.c:
10384 examples: fix include
10386 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10388 * gst/rtsp-server/rtsp-server.h:
10389 server: remove unused include
10391 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10393 * tests/check/Makefile.am:
10394 * tests/check/gst/mountpoints.c:
10395 tests: add test for mountpoints
10397 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10399 * gst/rtsp-server/rtsp-client.c:
10400 client: fix factory leak
10401 Keep the factory in the state object only for authorization checks and make
10402 sure we unref it on failure. Also don't keep invalid objects in the state
10405 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10407 * gst/rtsp-server/rtsp-mount-points.c:
10408 mounts: add g_return_if guards
10410 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10412 * tests/check/gst/client.c:
10413 tests: add more tests
10415 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10417 * gst/rtsp-server/rtsp-client.c:
10418 client: improve debug
10420 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10422 * gst/rtsp-server/rtsp-client.c:
10423 client: improve debug and fix leaks
10424 Cleanup the uri and session when there is a bad request.
10426 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10431 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10433 * tests/check/gst/client.c:
10434 test: add test for session in options request
10436 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10438 * gst/rtsp-server/rtsp-client.c:
10439 client: use 454 when session can't be found
10440 We should use 454 when a session can't be found because there was no session
10441 pool configured in the server. This is not a server configuration problem
10442 because the server on which the request is done might not be the same one that
10443 will keep the sessions for us and so it does not need to support sessions.
10445 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10447 * gst/rtsp-server/rtsp-client.c:
10448 client: only free connection when there is one
10449 It's possible that the client doesn't have a connection when we try to free it.
10451 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10453 * tests/check/Makefile.am:
10454 * tests/check/gst/client.c:
10455 tests: add unit test for the client object
10457 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10459 * gst/rtsp-server/rtsp-client.c:
10460 client: small cleanup
10462 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10464 * gst/rtsp-server/rtsp-client.h:
10465 client: remove unused include
10467 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10469 * gst/rtsp-server/rtsp-client.c:
10470 client: fix compilation
10472 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10474 * gst/rtsp-server/rtsp-client.c:
10475 client: call destroy without the lock
10477 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10479 * gst/rtsp-server/rtsp-client.c:
10480 * gst/rtsp-server/rtsp-client.h:
10481 client: make the client usable without a socket
10482 Make a method to let the client handle a message and a callback when the client
10483 wants us to send a response message back. This makes it possible to also use the
10484 client object without the sockets, which should make it easier to test.
10486 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10488 * gst/rtsp-server/rtsp-client.c:
10489 * gst/rtsp-server/rtsp-client.h:
10490 client: small cleanup
10492 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10494 * docs/libs/gst-rtsp-server-sections.txt:
10495 * gst/rtsp-server/rtsp-client.c:
10496 * gst/rtsp-server/rtsp-client.h:
10497 * gst/rtsp-server/rtsp-server.c:
10498 client: remove reference to server
10499 We don't need to keep a ref to the server
10501 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10503 * gst/rtsp-server/rtsp-client.c:
10504 * gst/rtsp-server/rtsp-client.h:
10505 client: add locking
10506 Also add some g_return_if()
10508 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10510 * gst/rtsp-server/rtsp-client.c:
10511 client: log more errors
10513 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10515 * gst/rtsp-server/rtsp-client.c:
10516 client: fix compilation
10518 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10520 * gst/rtsp-server/rtsp-client.c:
10521 * gst/rtsp-server/rtsp-client.h:
10522 client: add generic close-after-send support
10523 Add a property to send_response() to close the connection after the response has
10524 been sent to the client.
10526 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10529 * docs/libs/gst-rtsp-server-docs.sgml:
10530 * docs/libs/gst-rtsp-server-sections.txt:
10531 * docs/libs/gst-rtsp-server.types:
10532 * examples/test-auth.c:
10533 * examples/test-launch.c:
10534 * examples/test-mp4.c:
10535 * examples/test-multicast.c:
10536 * examples/test-multicast2.c:
10537 * examples/test-ogg.c:
10538 * examples/test-readme.c:
10539 * examples/test-sdp.c:
10540 * examples/test-uri.c:
10541 * examples/test-video.c:
10542 * gst/rtsp-server/Makefile.am:
10543 * gst/rtsp-server/rtsp-auth.h:
10544 * gst/rtsp-server/rtsp-client.c:
10545 * gst/rtsp-server/rtsp-client.h:
10546 * gst/rtsp-server/rtsp-media-mapping.c:
10547 * gst/rtsp-server/rtsp-media-mapping.h:
10548 * gst/rtsp-server/rtsp-mount-points.c:
10549 * gst/rtsp-server/rtsp-mount-points.h:
10550 * gst/rtsp-server/rtsp-server.c:
10551 * gst/rtsp-server/rtsp-server.h:
10552 * gst/rtsp-server/rtsp-session-media.c:
10553 * gst/rtsp-server/rtsp-session-pool.c:
10554 * gst/rtsp-server/rtsp-session-pool.h:
10555 * tests/check/gst/rtspserver.c:
10556 MediaMapping -> MountPoints
10557 Describes better what the object manages.
10559 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10562 configure: bump required version of -base
10564 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10566 * gst/rtsp-server/rtsp-media.c:
10569 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10571 * gst/rtsp-server/rtsp-media.c:
10572 * gst/rtsp-server/rtsp-media.h:
10573 media: support more Range formats
10574 Use the new -base methods to convert the Range string into a seek start and stop
10577 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10579 * examples/test-launch.c:
10580 examples: fix whitespace
10582 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10584 * examples/test-auth.c:
10585 test-auth: add example of how to remove sessions
10586 Add an example of the session filter api.
10588 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10590 * examples/test-uri.c:
10591 test-uri: remove mapping example
10593 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10595 * examples/test-uri.c:
10596 test-uri: fix callback signature
10598 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10600 * gst/rtsp-server/rtsp-media-factory.c:
10601 factory: keep ref to factory while media active
10602 While the media from a factory is alive, keep a ref to the factory.
10603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10605 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * gst/rtsp-server/rtsp-media-factory-uri.c:
10608 factory-uri: add some debug
10610 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10612 * gst/rtsp-server/rtsp-stream.c:
10613 stream: set udp sources to PLAYING
10614 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10615 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10617 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10619 * gst/rtsp-server/rtsp-media-factory-uri.c:
10620 factory-uri: take ref to factory
10621 Take a ref to the factory that we place in our list.
10623 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10625 * tests/Makefile.am:
10626 * tests/test-reuse.c:
10627 test: add test for server reuse
10628 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10630 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10632 * gst/rtsp-server/rtsp-server.c:
10633 server: start and stop multiple times
10634 Stop listening on the RTSP port when the GSource is removed, so clients
10635 can't connect and the server can be started again.
10636 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10638 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10640 * gst/rtsp-server/rtsp-server.c:
10641 server: fix small leak
10643 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10645 * gst/rtsp-server/rtsp-media.c:
10646 media: unref source in finish_unprepare
10647 The source is created in prepare, unref it in finish_unprepare.
10648 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10650 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10652 * gst/rtsp-server/rtsp-client.c:
10653 * gst/rtsp-server/rtsp-media.c:
10654 rtsp-media: remove bus watch before finalizing
10655 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10656 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10657 the GDestroyNotify function.
10658 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10659 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10660 gst_rtsp_media_unprepare before unreffing the media.
10661 This way, the bus watch will be removed before the media is finalized.
10662 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10664 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10666 * gst/rtsp-server/rtsp-client.c:
10667 * gst/rtsp-server/rtsp-client.h:
10668 client: wait until the TEARDOWN response is sent to close the connection
10669 Responses can be sent async so we need to wait until the TEARDOWN response has
10670 been written before we close the connection to the client. This avoids the risk
10671 of writing/polling closed sockets.
10672 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10674 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10676 * gst/rtsp-server/rtsp-stream.c:
10677 rtsp-stream: plug socket leak
10678 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10680 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10683 Automatic update of common submodule
10684 From 6bb6951 to a72faea
10686 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10688 * gst/rtsp-server/rtsp-media-factory-uri.c:
10689 rtsp-server: don't use deprecated API
10691 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10693 * gst/rtsp-server/rtsp-client.c:
10694 rtsp-client: fix unused-but-set-variable compiler warning
10695 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10697 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10700 * docs/libs/gst-rtsp-server-sections.txt:
10701 * gst/rtsp-server/rtsp-client.c:
10704 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10706 * examples/Makefile.am:
10707 * examples/test-multicast2.c:
10708 examples: add another multicast example
10709 Add an example for how to configure separate multicast ranges for each media
10712 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10714 * examples/test-multicast.c:
10717 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10719 * gst/rtsp-server/rtsp-client.c:
10720 * gst/rtsp-server/rtsp-media.c:
10721 * gst/rtsp-server/rtsp-session-media.c:
10722 * gst/rtsp-server/rtsp-session-media.h:
10723 * gst/rtsp-server/rtsp-stream-transport.c:
10724 * gst/rtsp-server/rtsp-stream-transport.h:
10725 stream: use the address managed by the stream
10726 Use the address managed by the stream for multicast. This allows us to have 1
10727 multicast address for each stream.
10728 Because the address is now managed by the stream we don't have to pass it around
10730 Set the address pool on the streams.
10732 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10734 * gst/rtsp-server/rtsp-client.c:
10735 * gst/rtsp-server/rtsp-media.c:
10736 * gst/rtsp-server/rtsp-stream.c:
10737 rtsp: improve debug
10739 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10741 * gst/rtsp-server/rtsp-media.c:
10742 * gst/rtsp-server/rtsp-media.h:
10743 media: add signal for new streams
10744 This allows applications to listen for new streams and configure properties on
10745 them, like the address pool.
10747 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10749 * gst/rtsp-server/rtsp-media.c:
10750 media: configure address pool in new streams
10752 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10754 * gst/rtsp-server/rtsp-stream.c:
10755 * gst/rtsp-server/rtsp-stream.h:
10756 stream: add methods to deal with address pool
10757 Add methods to get and set the address pool for the stream
10758 Add method to allocate and get the multicast addresses for this stream.
10760 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10762 * docs/libs/gst-rtsp-server-sections.txt:
10763 * gst/rtsp-server/rtsp-media.c:
10764 * gst/rtsp-server/rtsp-media.h:
10765 media: remove MTU property
10766 It is a stream property
10768 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10770 * gst/rtsp-server/rtsp-client.c:
10771 client: set blocksize only on stream
10772 Set the blocksize only on the current stream.
10774 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10776 * gst/rtsp-server/rtsp-stream.c:
10777 stream: share src and sink sockets
10778 the allocated socket is in the used-socket property, not socket.
10780 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10782 * gst/rtsp-server/rtsp-address-pool.c:
10783 * gst/rtsp-server/rtsp-address-pool.h:
10784 * gst/rtsp-server/rtsp-client.c:
10785 * gst/rtsp-server/rtsp-session-media.c:
10786 * gst/rtsp-server/rtsp-session-media.h:
10787 * gst/rtsp-server/rtsp-stream-transport.c:
10788 * gst/rtsp-server/rtsp-stream-transport.h:
10789 * tests/check/gst/addresspool.c:
10790 rtsp: make address-pool return an address object
10791 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
10792 store more info in the structure and allows us to more easily return the address
10793 to the right pool when no longer needed.
10794 Pass the address to the StreamTransport so that we can return it to the pool
10795 when the stream transport is freed or changed.
10797 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10799 * examples/Makefile.am:
10800 * examples/test-multicast.c:
10801 examples: add multicast example
10802 Show how to set up the multicast address pool so that media can be
10803 server with multicast.
10805 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10807 * gst/rtsp-server/rtsp-client.c:
10808 * gst/rtsp-server/rtsp-media-factory.c:
10809 * gst/rtsp-server/rtsp-media-factory.h:
10810 * gst/rtsp-server/rtsp-media.c:
10811 * gst/rtsp-server/rtsp-media.h:
10812 rtsp: use AddressPool
10813 Remove the multicast_group property.
10814 Use the configured addresspool to allocate multicast addresses.
10816 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10818 * gst/rtsp-server/rtsp-address-pool.c:
10819 * gst/rtsp-server/rtsp-address-pool.h:
10820 address-pool: add clear method
10822 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10824 * gst/rtsp-server/rtsp-address-pool.c:
10825 address-pool: small cleanups
10827 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10829 * tests/check/Makefile.am:
10830 * tests/check/gst/addresspool.c:
10831 tests: add addresspool unit test
10833 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10835 * gst/rtsp-server/Makefile.am:
10836 * gst/rtsp-server/rtsp-address-pool.c:
10837 * gst/rtsp-server/rtsp-address-pool.h:
10838 address-pool: add object to manage multicast addresses
10839 Make an object that can manage a rage of multicast addresses and ports.
10841 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10843 * gst/rtsp-server/rtsp-server.c:
10844 server: set default max-threads property
10846 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10848 * gst/rtsp-server/rtsp-media.c:
10849 media: wait for concurrent _prepare
10850 If a prepare is busy, wait for the result.
10852 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10854 * gst/rtsp-server/rtsp-media.c:
10855 media: add lock around message handler
10856 We don't want to dispatch messages while we are still processing the result of
10859 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10861 * gst/rtsp-server/rtsp-media.c:
10862 * gst/rtsp-server/rtsp-media.h:
10863 media: add lock to protect state changes
10865 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10867 * gst/rtsp-server/rtsp-stream.c:
10868 * gst/rtsp-server/rtsp-stream.h:
10869 stream: add locking
10871 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10873 * gst/rtsp-server/rtsp-stream-transport.c:
10874 * gst/rtsp-server/rtsp-stream-transport.h:
10875 * gst/rtsp-server/rtsp-stream.c:
10876 stream-transport: add keep-alive method
10878 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10880 * gst/rtsp-server/rtsp-stream-transport.c:
10881 * gst/rtsp-server/rtsp-stream-transport.h:
10882 * gst/rtsp-server/rtsp-stream.c:
10883 stream-transport: add method to handle RTP/RTCP
10884 Call new methods instead of poking into the structures directly.
10886 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10888 * gst/rtsp-server/rtsp-session-media.c:
10889 * gst/rtsp-server/rtsp-session-media.h:
10890 session-media: add locking
10892 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10894 * gst/rtsp-server/rtsp-session.c:
10895 * gst/rtsp-server/rtsp-session.h:
10896 session: add locking
10898 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10900 * gst/rtsp-server/rtsp-server.c:
10901 server: free old socket
10903 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10905 * gst/rtsp-server/rtsp-media-mapping.c:
10906 * gst/rtsp-server/rtsp-media-mapping.h:
10907 mapping: add locking
10909 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10911 * gst/rtsp-server/rtsp-media-factory.c:
10912 media-factory: add locking
10914 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10916 * gst/rtsp-server/rtsp-auth.c:
10917 * gst/rtsp-server/rtsp-auth.h:
10920 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10922 * gst/rtsp-server/rtsp-server.c:
10923 * gst/rtsp-server/rtsp-server.h:
10924 server: add max-thread property
10926 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10928 * gst/rtsp-server/rtsp-server.c:
10929 * gst/rtsp-server/rtsp-server.h:
10930 server: use a threadpool for the mainloops
10932 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10934 * gst/rtsp-server/rtsp-client.c:
10935 * gst/rtsp-server/rtsp-client.h:
10936 client: rename method
10937 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
10938 don't really create the client from the socket, we use the socket for the
10941 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10943 * gst/rtsp-server/rtsp-client.c:
10944 * gst/rtsp-server/rtsp-client.h:
10945 * gst/rtsp-server/rtsp-server.c:
10946 server: rework maincontext handling in clients
10947 Make a separate method to attach a client to a MainContext.
10948 Let the server decide in what GMainContext the client will operate and give this
10949 context to the client in attach. Then the server can later decide to use a
10950 separate thread for each client or just use the mainthread.
10952 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10954 * gst/rtsp-server/rtsp-client.c:
10955 * gst/rtsp-server/rtsp-session.c:
10956 * gst/rtsp-server/rtsp-session.h:
10957 session: move session header code in session object
10959 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10963 * examples/test-auth.c:
10964 * examples/test-launch.c:
10965 * examples/test-mp4.c:
10966 * examples/test-ogg.c:
10967 * examples/test-readme.c:
10968 * examples/test-sdp.c:
10969 * examples/test-uri.c:
10970 * examples/test-video.c:
10971 * gst/rtsp-server/rtsp-auth.c:
10972 * gst/rtsp-server/rtsp-auth.h:
10973 * gst/rtsp-server/rtsp-client.c:
10974 * gst/rtsp-server/rtsp-client.h:
10975 * gst/rtsp-server/rtsp-media-factory-uri.c:
10976 * gst/rtsp-server/rtsp-media-factory-uri.h:
10977 * gst/rtsp-server/rtsp-media-factory.c:
10978 * gst/rtsp-server/rtsp-media-factory.h:
10979 * gst/rtsp-server/rtsp-media-mapping.c:
10980 * gst/rtsp-server/rtsp-media-mapping.h:
10981 * gst/rtsp-server/rtsp-media.c:
10982 * gst/rtsp-server/rtsp-media.h:
10983 * gst/rtsp-server/rtsp-params.c:
10984 * gst/rtsp-server/rtsp-params.h:
10985 * gst/rtsp-server/rtsp-sdp.c:
10986 * gst/rtsp-server/rtsp-sdp.h:
10987 * gst/rtsp-server/rtsp-server.c:
10988 * gst/rtsp-server/rtsp-server.h:
10989 * gst/rtsp-server/rtsp-session-media.c:
10990 * gst/rtsp-server/rtsp-session-media.h:
10991 * gst/rtsp-server/rtsp-session-pool.c:
10992 * gst/rtsp-server/rtsp-session-pool.h:
10993 * gst/rtsp-server/rtsp-session.c:
10994 * gst/rtsp-server/rtsp-session.h:
10995 * gst/rtsp-server/rtsp-stream-transport.c:
10996 * gst/rtsp-server/rtsp-stream-transport.h:
10997 * gst/rtsp-server/rtsp-stream.c:
10998 * gst/rtsp-server/rtsp-stream.h:
10999 * tests/check/gst/rtspserver.c:
11000 * tests/test-cleanup.c:
11003 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11005 * gst/rtsp-server/rtsp-media.c:
11006 * gst/rtsp-server/rtsp-session-media.c:
11007 * gst/rtsp-server/rtsp-session.c:
11008 rtsp-server: added annotations to indicate type of ownership transfer of return values
11009 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11011 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11014 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11016 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11019 * bindings/Makefile.am:
11020 * bindings/vala/Makefile.am:
11021 * bindings/vala/gst-rtsp-server-0.10.deps:
11022 * bindings/vala/gst-rtsp-server-0.10.vapi:
11023 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11024 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11025 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11026 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11027 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11029 bindings: remove vala bindings
11030 They'll be reunited with the other GStreamer bindings
11031 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11033 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11035 * gst/rtsp-server/rtsp-client.c:
11036 * gst/rtsp-server/rtsp-session-media.c:
11037 * gst/rtsp-server/rtsp-session-media.h:
11038 * gst/rtsp-server/rtsp-stream-transport.c:
11039 * gst/rtsp-server/rtsp-stream-transport.h:
11040 rtsp: only create transport when needed
11041 Only create the StreamTransport when configured.
11043 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11045 * gst/rtsp-server/rtsp-client.c:
11046 client: small cleanup
11048 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11050 * gst/rtsp-server/rtsp-client.c:
11051 * gst/rtsp-server/rtsp-client.h:
11052 * gst/rtsp-server/rtsp-stream-transport.c:
11053 * gst/rtsp-server/rtsp-stream-transport.h:
11054 rtsp: refactor configuration of transport
11055 Move the configuration of the transport to a place where it makes
11058 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11060 * gst/rtsp-server/rtsp-client.c:
11061 client: refactor transport parsing
11063 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11065 * gst/rtsp-server/rtsp-client.c:
11066 client: refuse to change the MTU on shared media
11067 If we change the MTU of chared media, it changes for all clients.
11068 We don't want to set the MTU to something large for clients that
11071 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11073 * examples/test-mp4.c:
11074 * gst/rtsp-server/rtsp-media.c:
11075 small fixes to docs and debug
11077 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11079 * gst/rtsp-server/rtsp-stream.c:
11080 stream: transports must already have been removed
11082 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11084 * gst/rtsp-server/rtsp-media.c:
11085 * gst/rtsp-server/rtsp-stream.c:
11086 * gst/rtsp-server/rtsp-stream.h:
11087 stream: improve join and leave of the pipeline
11089 Do the cleanup properly
11092 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11094 * gst/rtsp-server/rtsp-media.c:
11095 media: move unprepare below default implementation
11096 Makes it easier to find the default implementation
11098 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11100 * gst/rtsp-server/rtsp-media.c:
11101 media: signal unprepared when we actually finish
11103 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11105 * gst/rtsp-server/rtsp-media.c:
11106 media: no need to unlock, unprepare does that when needed
11108 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11110 * docs/libs/gst-rtsp-server-sections.txt:
11111 * gst/rtsp-server/rtsp-media-factory.h:
11112 * gst/rtsp-server/rtsp-media-mapping.c:
11113 * gst/rtsp-server/rtsp-media.h:
11114 * gst/rtsp-server/rtsp-params.c:
11115 * gst/rtsp-server/rtsp-server.c:
11116 * gst/rtsp-server/rtsp-session-pool.h:
11117 * gst/rtsp-server/rtsp-session.c:
11118 * gst/rtsp-server/rtsp-session.h:
11119 * gst/rtsp-server/rtsp-stream-transport.h:
11120 * gst/rtsp-server/rtsp-stream.h:
11123 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11125 * gst/rtsp-server/rtsp-client.c:
11126 * gst/rtsp-server/rtsp-media-mapping.h:
11127 * gst/rtsp-server/rtsp-media.c:
11128 * gst/rtsp-server/rtsp-media.h:
11129 * gst/rtsp-server/rtsp-server.h:
11130 * gst/rtsp-server/rtsp-stream.c:
11131 * gst/rtsp-server/rtsp-stream.h:
11132 rtsp: fix MTU setting
11133 Fix setting of the MTU. There is no need for a vmethod.
11135 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11140 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11143 configure: bump version number after refactoring
11145 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11147 * gst/rtsp-server/Makefile.am:
11148 * gst/rtsp-server/rtsp-client.c:
11149 * gst/rtsp-server/rtsp-client.h:
11150 * gst/rtsp-server/rtsp-media-factory-uri.c:
11151 * gst/rtsp-server/rtsp-media-factory.c:
11152 * gst/rtsp-server/rtsp-media-factory.h:
11153 * gst/rtsp-server/rtsp-media.c:
11154 * gst/rtsp-server/rtsp-media.h:
11155 * gst/rtsp-server/rtsp-sdp.c:
11156 * gst/rtsp-server/rtsp-session-media.c:
11157 * gst/rtsp-server/rtsp-session-media.h:
11158 * gst/rtsp-server/rtsp-session.c:
11159 * gst/rtsp-server/rtsp-session.h:
11160 * gst/rtsp-server/rtsp-stream-transport.c:
11161 * gst/rtsp-server/rtsp-stream-transport.h:
11162 * gst/rtsp-server/rtsp-stream.c:
11163 * gst/rtsp-server/rtsp-stream.h:
11164 rtsp: massive refactoring
11165 Make GObjects from the remaining simple structures.
11166 Remove GstRTSPSessionStream, it's not needed.
11167 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11168 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11169 a GstRTSPStream should be transported to a client.
11170 Rename GstRTSPMediaFactory::get_element -> create_element because that
11171 more accurately describes what it does.
11172 Make nice methods instead of poking in the structures.
11173 Move some methods inside the relevant object source code.
11174 Use GPtrArray to store objects instead of plain arrays, it is more
11175 natural and allows us to more easily clean up.
11176 Move the allocation of udp ports to the Stream object. The Stream object
11177 contains the elements needed to stream the media to a client.
11178 Improve the prepare and unprepare methods. Unprepare should now undo
11179 everything prepare did. Improve also async unprepare when doing EOS on
11180 shutdown. Make sure we always unprepare correctly.
11182 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11184 * gst/rtsp-server/rtsp-client.c:
11185 rtsp-client: Unref server address clients connected to
11186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11188 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11190 * gst/rtsp-server/rtsp-server.c:
11191 rtsp-server: don't ref server socket if it is NULL
11192 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11193 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11195 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11197 * tests/check/Makefile.am:
11198 tests: Add libgio link dependency
11199 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11201 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11203 * gst/rtsp-server/rtsp-media-mapping.c:
11204 * gst/rtsp-server/rtsp-media-mapping.h:
11205 rtsp-media-mapping: rename find_media vfunc to find_factory
11206 The virtual method and class method should have the same name
11207 so it is correctly represented in GIR file
11208 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11210 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11212 * gst/rtsp-server/rtsp-auth.c:
11213 * gst/rtsp-server/rtsp-client.c:
11214 * gst/rtsp-server/rtsp-media-factory-uri.c:
11215 * gst/rtsp-server/rtsp-media-factory.c:
11216 * gst/rtsp-server/rtsp-media-mapping.c:
11217 * gst/rtsp-server/rtsp-media.c:
11218 * gst/rtsp-server/rtsp-server.c:
11219 * gst/rtsp-server/rtsp-session-pool.c:
11220 * gst/rtsp-server/rtsp-session.c:
11221 rtsp-server: fixed comments and GIR annotations
11222 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11224 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11226 * gst/rtsp-server/rtsp-media-mapping.c:
11227 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11229 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11231 * gst/rtsp-server/rtsp-server.c:
11232 rtsp-server: allow binding on port 0 (binds on a random port)
11234 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11236 * gst/rtsp-server/rtsp-server.c:
11237 * gst/rtsp-server/rtsp-server.h:
11238 rtsp-server: add bound-port property
11239 bound-port can be used to retrieve the port number when the server is bound on
11240 port 0, which binds on a random port.
11242 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11244 * gst/rtsp-server/rtsp-media-factory.c:
11245 * gst/rtsp-server/rtsp-media-factory.h:
11246 rtsp-media-factory: make ::get_element overridable by GI bindings
11247 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11248 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11249 as the invoker for ::get_element(), making it overridable by GI generated
11252 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11254 * gst/rtsp-server/rtsp-media-factory-uri.c:
11255 rtsp-media-factory-uri: don't autoplug parsers in a loop
11256 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11259 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11261 * gst/rtsp-server/Makefile.am:
11262 Explicitly link against gio. Fix link error on mac.
11264 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11266 * gst/rtsp-server/rtsp-session.c:
11267 session: add ttl to the transport header in SETUP
11268 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11270 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11272 * gst/rtsp-server/rtsp-client.c:
11273 * gst/rtsp-server/rtsp-client.h:
11274 * gst/rtsp-server/rtsp-media.c:
11275 client: Use client transport settings for multicast if allowed.
11276 This patch makes it possible for the client to send transport settings for
11277 multicast (destination && ttl). Client settings must be explicitly allowed or
11278 the server will use its own settings.
11279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11281 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11284 Automatic update of common submodule
11285 From 6c0b52c to 6bb6951
11287 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11289 * gst/rtsp-server/rtsp-client.c:
11290 rtsp-client: do not destroy the rtsp watch
11291 Don't destroy the client watch while dispatching. The rtsp watch is
11292 automatically destroyed after the rtsp watch function closed() has
11294 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11296 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11299 Automatic update of common submodule
11300 From 4f962f7 to 6c0b52c
11302 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11304 * gst/rtsp-server/rtsp-media.c:
11305 media: fix check for seekability
11307 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11309 * gst/rtsp-server/rtsp-client.c:
11310 client: use more GIO
11311 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11313 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11315 * gst/rtsp-server/rtsp-server.c:
11316 server: remove obsolete includes
11318 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11320 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11321 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11322 be available in "on_new_ssrc". The transports are added in
11323 gst_rtsp_media_set_state when going to PLAYING state. However,
11324 "on_new_ssrc" might be called before this happens.
11325 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11327 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11329 * gst/rtsp-server/rtsp-client.c:
11330 * gst/rtsp-server/rtsp-client.h:
11331 rtsp-client: add signals for rtsp requests (fixes #683287)
11333 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11335 * gst/rtsp-server/rtsp-client.c:
11336 * gst/rtsp-server/rtsp-client.h:
11337 add new-session signal to rtsp-client (fixes #683058)
11339 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11342 Automatic update of common submodule
11343 From 668acee to 4f962f7
11345 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11347 * gst/rtsp-server/rtsp-server.c:
11348 * tests/check/gst/rtspserver.c:
11349 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11350 Do not assume that *error is set in g_socket_address_enumerator_next.
11351 Added test_bind_already_in_use unit-test.
11352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11354 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11357 Automatic update of common submodule
11358 From 94ccf4c to 668acee
11360 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11362 * gst/rtsp-server/rtsp-client.c:
11363 * gst/rtsp-server/rtsp-client.h:
11364 rtsp-client: make create_sdp virtual method
11365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11367 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11370 Automatic update of common submodule
11371 From 98e386f to 94ccf4c
11373 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11375 * gst/rtsp-server/rtsp-client.c:
11378 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11380 * gst/rtsp-server/rtsp-client.c:
11381 * gst/rtsp-server/rtsp-client.h:
11382 * gst/rtsp-server/rtsp-server.c:
11383 * gst/rtsp-server/rtsp-server.h:
11384 rtsp-server: use an existing socket to establish HTTP tunnel
11385 Make it possible to transfer a socket from an HTTP server to be used as
11386 an RTSP over HTTP tunnel.
11388 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11390 * gst/rtsp-server/rtsp-client.c:
11391 * gst/rtsp-server/rtsp-media.c:
11392 * gst/rtsp-server/rtsp-media.h:
11393 rtsp: Handle the blocksize parameter
11394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11396 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11398 * tests/check/Makefile.am:
11399 * tests/check/gst/rtspserver.c:
11400 Have unit test get header from source dir, not installed dir
11401 This makes compilation of unit tests work in a build directory other
11402 than the source directory.
11403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11405 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11407 * gst/rtsp-server/rtsp-media.c:
11408 rtsp-media: update for gst_element_make_from_uri() changes
11410 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11413 * tests/Makefile.am:
11414 * tests/check/Makefile.am:
11415 * tests/check/gst/rtspserver.c:
11416 rtsp: add unit test
11417 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11419 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11421 * gst/rtsp-server/rtsp-media.c:
11422 rtsp-media: don't collect media stats when going to NULL
11423 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11425 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11427 * gst/rtsp-server/rtsp-client.c:
11428 client: don't leak transports
11430 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11432 * gst/rtsp-server/rtsp-client.c:
11433 rtsp-client: free transport on no_stream in SETUP handler
11435 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11437 * gst/rtsp-server/rtsp-client.c:
11438 rtsp-client: changed session media iteration
11439 In client_unlink_session: now don't iterate in session->medias
11440 list where items are removed by gst_rtsp_session_release_media.
11441 Instead, repeatedly remove the first item.
11443 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11445 * gst/rtsp-server/rtsp-client.c:
11446 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11447 GstRTSPSessionMedia is not a GObject type. When the
11448 GstRTSPSession is freed, it will free the media.
11450 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11452 * gst/rtsp-server/rtsp-media-factory.c:
11453 factory: plug pad leak in collect_streams
11454 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11455 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11456 will take one reference, and the other reference will otherwise
11457 give a memory leak.
11459 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11462 configure: suppress some warnings when debug is disabled
11463 Warnings about unused variables should be suppressed if core has the
11464 debug system disabled.
11465 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11467 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11469 * docs/libs/Makefile.am:
11470 docs: fix build in uninstalled setup
11471 Include gst-plugins-base libs properly.
11473 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11475 * docs/libs/gst-rtsp-server.types:
11476 docs: include headers defining rtsp-server object types
11477 Fixes compiler warnings during docs build.
11478 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11480 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11483 configure: Add warning flags for compiler when configuring
11484 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11486 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11489 Automatic update of common submodule
11490 From 03a0e57 to 98e386f
11492 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11495 Automatic update of common submodule
11496 From 1fab359 to 03a0e57
11498 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11500 * gst/rtsp-server/rtsp-client.c:
11501 client: fix GSocketAddress leak in gst_rtsp_client_accept
11502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11504 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11507 Automatic update of common submodule
11508 From f1b5a96 to 1fab359
11510 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11513 Automatic update of common submodule
11514 From 92b7266 to f1b5a96
11516 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11519 Automatic update of common submodule
11520 From ec1c4a8 to 92b7266
11522 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11525 Automatic update of common submodule
11526 From 3429ba6 to ec1c4a8
11528 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11530 * gst/rtsp-server/rtsp-auth.c:
11531 * gst/rtsp-server/rtsp-client.c:
11532 * gst/rtsp-server/rtsp-media-factory-uri.c:
11533 * gst/rtsp-server/rtsp-server.c:
11534 rtsp: fix compiler warnings
11535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11537 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11540 Automatic update of common submodule
11541 From dc70203 to 3429ba6
11543 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11545 * gst/rtsp-server/rtsp-client.c:
11546 * gst/rtsp-server/rtsp-media-factory.c:
11547 * gst/rtsp-server/rtsp-media-factory.h:
11548 * gst/rtsp-server/rtsp-media.c:
11549 * gst/rtsp-server/rtsp-media.h:
11550 * gst/rtsp-server/rtsp-server.c:
11551 * gst/rtsp-server/rtsp-server.h:
11552 * gst/rtsp-server/rtsp-session-pool.c:
11553 * gst/rtsp-server/rtsp-session-pool.h:
11554 rtsp-server: port to new thread API
11556 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11559 Automatic update of common submodule
11560 From 6db25be to dc70203
11562 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11564 * gst/rtsp-server/rtsp-auth.c:
11565 * gst/rtsp-server/rtsp-auth.h:
11566 * gst/rtsp-server/rtsp-client.c:
11567 rtsp-server: Fix compilation and compiler warnings
11569 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11573 * gst/rtsp-server/Makefile.am:
11574 configure: Modernize autotools setup a bit
11575 Also we now only create tar.bz2 and tar.xz tarballs.
11577 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11580 Automatic update of common submodule
11581 From 464fe15 to 6db25be
11583 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11586 Automatic update of common submodule
11587 From 7fda524 to 464fe15
11589 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11592 * docs/libs/Makefile.am:
11593 * docs/version.entities.in:
11594 * gst-rtsp.spec.in:
11595 * gst/rtsp-server/Makefile.am:
11596 * pkgconfig/Makefile.am:
11597 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11598 * pkgconfig/gstreamer-rtsp-server.pc.in:
11599 * tests/Makefile.am:
11600 rtsp-server: Update versioning
11602 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11604 Merge remote-tracking branch 'origin/0.10'
11606 gst/rtsp-server/rtsp-session-pool.c
11608 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11610 * gst/rtsp-server/rtsp-session-pool.c:
11611 rtsp-server: Don't use deprecated GLib API
11613 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11615 Replace master with 0.11
11617 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11619 Merge branch 'master' into 0.11
11621 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11623 Merge branch 'master' into 0.11
11625 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11628 A couple minor typo fixes
11630 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11632 * gst/rtsp-server/rtsp-media.c:
11633 media: fix state of the appqueue
11635 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11637 * gst/rtsp-server/rtsp-media-factory-uri.c:
11638 factory: use videoconvert
11640 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11642 * gst/rtsp-server/rtsp-media-factory-uri.c:
11643 factory: change to new style caps
11645 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11647 * gst/rtsp-server/rtsp-client.c:
11648 * gst/rtsp-server/rtsp-client.h:
11649 * gst/rtsp-server/rtsp-media-factory-uri.c:
11650 * gst/rtsp-server/rtsp-media.c:
11651 * gst/rtsp-server/rtsp-server.c:
11652 * gst/rtsp-server/rtsp-server.h:
11653 * gst/rtsp-server/rtsp-session-pool.c:
11654 rtsp-server: port to GIO
11657 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11660 configure: fix build
11662 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11665 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11666 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11668 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11671 * examples/Makefile.am:
11672 First rule of gst-rtsp-server club: don't talk about gst-phonon
11674 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11677 * pkgconfig/Makefile.am:
11678 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11679 * pkgconfig/gstreamer-rtsp-server.pc.in:
11680 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11681 For consistency with all other modules.
11683 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11685 * gst/rtsp-server/rtsp-client.c:
11686 rtsp-client: update for new map API
11688 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11691 * bindings/Makefile.am:
11692 * bindings/python/Makefile.am:
11693 * bindings/python/arg-types.py:
11694 * bindings/python/codegen/Makefile.am:
11695 * bindings/python/codegen/__init__.py:
11696 * bindings/python/codegen/argtypes.py:
11697 * bindings/python/codegen/code-coverage.py:
11698 * bindings/python/codegen/codegen.py:
11699 * bindings/python/codegen/definitions.py:
11700 * bindings/python/codegen/defsparser.py:
11701 * bindings/python/codegen/docextract.py:
11702 * bindings/python/codegen/docgen.py:
11703 * bindings/python/codegen/fileprefix.override:
11704 * bindings/python/codegen/fileprefixmodule.c:
11705 * bindings/python/codegen/h2def.py:
11706 * bindings/python/codegen/mergedefs.py:
11707 * bindings/python/codegen/mkskel.py:
11708 * bindings/python/codegen/override.py:
11709 * bindings/python/codegen/reversewrapper.py:
11710 * bindings/python/codegen/scmexpr.py:
11711 * bindings/python/rtspserver-types.defs:
11712 * bindings/python/rtspserver.defs:
11713 * bindings/python/rtspserver.override:
11714 * bindings/python/rtspservermodule.c:
11715 * bindings/python/test.py:
11717 python: remove pygst-based python bindings
11718 pygi is the future, apparently.
11720 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
11723 Automatic update of common submodule
11724 From c463bc0 to 7fda524
11726 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11729 Automatic update of common submodule
11730 From 2a59016 to c463bc0
11732 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11735 Automatic update of common submodule
11736 From 0807187 to 2a59016
11738 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11741 Automatic update of common submodule
11742 From 11f0cd5 to 0807187
11744 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11746 * examples/test-auth.c:
11747 example: update for new caps
11749 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11751 * examples/test-video.c:
11752 * gst/rtsp-server/rtsp-client.c:
11753 * gst/rtsp-server/rtsp-media-factory-uri.c:
11754 * gst/rtsp-server/rtsp-media.c:
11755 * gst/rtsp-server/rtsp-media.h:
11756 * gst/rtsp-server/rtsp-session.c:
11757 * gst/rtsp-server/rtsp-session.h:
11758 rtsp-server: port some more to 0.11
11760 Remove bufferlist stuff
11761 Update for new API.
11762 Add queue before appsink now that preroll-queue-len is gone.
11763 Update for request pad changes.
11765 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11767 Merge branch 'master' into 0.11
11769 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11771 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11772 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11773 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11775 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11777 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11778 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11779 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11781 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11783 Merge branch 'master' into 0.11
11785 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11787 * gst/rtsp-server/rtsp-media.c:
11788 * gst/rtsp-server/rtsp-media.h:
11789 media: add a seekable boolean
11790 Maintain the seekable state with a new variable instead of reusing the
11793 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
11795 * gst/rtsp-server/rtsp-media.c:
11796 Disallow seek in live media
11798 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11800 Merge branch 'master' into 0.11
11802 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
11804 * gst/rtsp-server/rtsp-server.c:
11805 #ifdef statements for windows socket creation were missing
11807 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
11810 Automatic update of common submodule
11811 From a39eb83 to 11f0cd5
11813 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
11816 Automatic update of common submodule
11817 From 605cd9a to a39eb83
11819 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11821 Merge branch 'master' into 0.11
11823 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11825 * gst/rtsp-server/rtsp-client.c:
11826 client: use method to access property
11828 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11830 * gst/rtsp-server/rtsp-media-factory.c:
11831 * gst/rtsp-server/rtsp-media-factory.h:
11832 media-factory: add protocols property
11833 Add a property to configure the allowed protocols in the media created from the
11836 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11838 * gst/rtsp-server/rtsp-media-factory.c:
11839 * gst/rtsp-server/rtsp-media-factory.h:
11840 media-factory: add media-configure signal
11841 Add signal to allow the application to configure the media after it was created
11844 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11846 * gst/rtsp-server/rtsp-client.c:
11847 client: use method to access property
11849 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11851 * gst/rtsp-server/rtsp-media-factory.c:
11852 * gst/rtsp-server/rtsp-media-factory.h:
11853 media-factory: add protocols property
11854 Add a property to configure the allowed protocols in the media created from the
11857 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11859 * gst/rtsp-server/rtsp-media-factory.c:
11860 * gst/rtsp-server/rtsp-media-factory.h:
11861 media-factory: add media-configure signal
11862 Add signal to allow the application to configure the media after it was created
11865 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11867 Merge branch 'master' into 0.11
11869 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11871 * gst/rtsp-server/rtsp-client.c:
11872 client: use media multicast group
11874 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11876 * gst/rtsp-server/rtsp-media-factory.h:
11877 * gst/rtsp-server/rtsp-server.h:
11878 * gst/rtsp-server/rtsp-session-pool.h:
11879 * gst/rtsp-server/rtsp-session.h:
11882 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11884 * gst/rtsp-server/rtsp-client.c:
11885 * gst/rtsp-server/rtsp-sdp.h:
11886 sdp: copy and free the server ip address
11887 Copy and free the server ip address to make memory management easier later.
11889 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11891 * gst/rtsp-server/rtsp-media-factory.c:
11892 media-factory: configure multicast in media
11894 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11896 * gst/rtsp-server/rtsp-media.c:
11897 * gst/rtsp-server/rtsp-media.h:
11898 media: add property for multicast group
11899 Add a property to configure the multicast group in the media.
11900 Based on patches from Marc Leeman and Robert Krakora.
11902 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11904 * gst/rtsp-server/rtsp-media-factory.c:
11905 * gst/rtsp-server/rtsp-media-factory.h:
11906 media-factory: add property for multicast group
11907 Add a property to configure the multicast group in the media factory.
11908 Based on patches from Marc Leeman and Robert Krakora.
11910 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11912 * gst/rtsp-server/rtsp-client.c:
11913 client: do configuration of transport in one place
11914 Move the configuration of the transport destination address to where we also
11915 configure the other bits.
11917 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11919 * gst/rtsp-server/rtsp-client.c:
11920 client: use media multicast group
11922 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11924 * gst/rtsp-server/rtsp-media-factory.h:
11925 * gst/rtsp-server/rtsp-server.h:
11926 * gst/rtsp-server/rtsp-session-pool.h:
11927 * gst/rtsp-server/rtsp-session.h:
11930 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11932 * gst/rtsp-server/rtsp-client.c:
11933 * gst/rtsp-server/rtsp-sdp.h:
11934 sdp: copy and free the server ip address
11935 Copy and free the server ip address to make memory management easier later.
11937 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11939 * gst/rtsp-server/rtsp-media-factory.c:
11940 media-factory: configure multicast in media
11942 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11944 * gst/rtsp-server/rtsp-media.c:
11945 * gst/rtsp-server/rtsp-media.h:
11946 media: add property for multicast group
11947 Add a property to configure the multicast group in the media.
11948 Based on patches from Marc Leeman and Robert Krakora.
11950 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11952 * gst/rtsp-server/rtsp-media-factory.c:
11953 * gst/rtsp-server/rtsp-media-factory.h:
11954 media-factory: add property for multicast group
11955 Add a property to configure the multicast group in the media factory.
11956 Based on patches from Marc Leeman and Robert Krakora.
11958 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11960 * gst/rtsp-server/rtsp-client.c:
11961 client: do configuration of transport in one place
11962 Move the configuration of the transport destination address to where we also
11963 configure the other bits.
11965 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11967 Merge branch 'master' into 0.11
11969 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11971 * gst/rtsp-server/rtsp-client.c:
11972 client: destroy pipeline on client disconnect with no prior TEARDOWN.
11973 The problem occurs when the client abruptly closes the connection without
11974 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
11975 server is where the pipeline gets torn down. Since this handler is not called,
11976 the pipeline remains and is up and running. Subsequent clients get their own
11977 pipelines and if the do not issue TEARDOWNs then those pipelines will also
11978 remain up and running. This is a resource leak.
11980 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11982 Merge branch 'master' into 0.11
11984 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
11986 * gst/rtsp-server/rtsp-media-factory.c:
11987 * gst/rtsp-server/rtsp-media-factory.h:
11988 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
11989 For example, it can be used to retrieve source elements like appsrc, in a more
11990 convenient way than subclassing get_element.
11992 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11994 Merge branch 'master' into 0.11
11996 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
11998 * gst/rtsp-server/rtsp-server.c:
11999 rtsp-server: hold on to reference while using object
12001 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12003 * gst/rtsp-server/rtsp-media.c:
12006 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12009 configure: use unstable api
12011 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12013 * gst/rtsp-server/rtsp-client.c:
12014 client: fix reference counting
12016 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12018 * gst/rtsp-server/rtsp-client.c:
12019 * gst/rtsp-server/rtsp-media.c:
12020 fix compiler warnings about unused variables
12022 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12024 * examples/test-launch.c:
12025 * examples/test-readme.c:
12026 * examples/test-uri.c:
12027 * examples/test-video.c:
12028 examples: tell rtsp uri when ready
12030 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12033 Automatic update of common submodule
12034 From 69b981f to 605cd9a
12036 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12038 * gst/rtsp-server/rtsp-client.c:
12039 client: update for buffer API change
12041 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12043 * gst/rtsp-server/Makefile.am:
12044 Makefile.am: 0.10 => @GST_MAJORMINOR@
12046 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12048 * gst/rtsp-server/rtsp-media-factory-uri.c:
12049 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12051 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12053 * gst/rtsp-server/.gitignore:
12054 .gitignore: 0.10 => 0.11
12056 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12058 * gst/rtsp-server/Makefile.am:
12059 Makefile.am: 0.10 => @GST_MAJORMINOR@
12061 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12063 Merge branch 'master' into 0.11
12065 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12068 Automatic update of common submodule
12069 From 9e5bbd5 to 69b981f
12071 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12074 Automatic update of common submodule
12075 From fd35073 to 9e5bbd5
12077 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12080 Automatic update of common submodule
12081 From 46dfcea to fd35073
12083 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12085 * gst/rtsp-server/rtsp-media-factory-uri.c:
12086 * gst/rtsp-server/rtsp-media.c:
12087 media: port to new caps API
12089 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12091 Merge branch 'master' into 0.11
12093 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12095 * bindings/vala/gst-rtsp-server-0.10.vapi:
12096 Updated Vala bindings.
12097 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12099 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12101 * gst/rtsp-server/rtsp-server.c:
12102 * gst/rtsp-server/rtsp-server.h:
12103 Add a signal for newly connected clients.
12104 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12106 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12108 * bindings/python/rtspserver.override:
12109 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12111 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12113 * gst/rtsp-server/Makefile.am:
12114 * gst/rtsp-server/rtsp-client.c:
12115 * gst/rtsp-server/rtsp-funnel.c:
12116 * gst/rtsp-server/rtsp-funnel.h:
12117 * gst/rtsp-server/rtsp-media.c:
12118 rtsp-server: port to 0.11
12120 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12125 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12127 Merge branch 'master' into 0.11
12132 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12135 Automatic update of common submodule
12136 From c3cafe1 to 46dfcea
12138 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12140 * bindings/python/Makefile.am:
12141 * bindings/python/rtspserver.defs:
12142 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12144 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12146 * bindings/python/arg-types.py:
12147 python bindings: add GstRTSPUrlParam
12148 Needed to implement MediaFactory virtual proxies
12150 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12152 * bindings/python/arg-types.py:
12153 python bindings: fix returning GstRTSPUrl types
12155 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12157 * bindings/python/arg-types.py:
12158 python bindings: add arg type for GstRTSPUrl
12160 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12162 * bindings/python/rtspserver.defs:
12163 python bindings: fix the definition of MediaFactory.collect_stream
12165 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12168 Automatic update of common submodule
12169 From 1ccbe09 to c3cafe1
12171 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12174 Automatic update of common submodule
12175 From 193b717 to 1ccbe09
12177 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12180 Automatic update of common submodule
12181 From b77e2bf to 193b717
12183 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12186 build: Include lcov.mak to allow test coverage report generation
12188 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12191 Automatic update of common submodule
12192 From d8814b6 to b77e2bf
12194 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12197 Automatic update of common submodule
12198 From 6aaa286 to d8814b6
12200 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12203 Automatic update of common submodule
12204 From 6aec6b9 to 6aaa286
12206 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12209 autogen: wingo signed comment
12211 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12213 * gst/rtsp-server/rtsp-session-pool.c:
12214 session: use full charset for RTSP session ID
12215 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12216 session ID more difficult.
12217 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12219 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12221 * gst/rtsp-server/Makefile.am:
12222 rtsp-server: Don't install the funnel header
12224 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12227 Automatic update of common submodule
12228 From 1de7f6a to 6aec6b9
12230 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12233 configure: require core/base 0.10.31
12234 Needed at least for gst_plugin_feature_rank_compare_func().
12236 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12239 Automatic update of common submodule
12240 From f94d739 to 1de7f6a
12242 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12244 * gst/rtsp-server/rtsp-media.c:
12245 media: remove more unused code
12247 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12249 * gst/rtsp-server/rtsp-media.c:
12250 * gst/rtsp-server/rtsp-media.h:
12251 media: remove duplicate filtering
12252 Remove the duplicate filtering code now that we have a released -good version.
12253 Give a warning instead.
12255 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12257 * gst/rtsp-server/rtsp-media-factory.c:
12258 * gst/rtsp-server/rtsp-media.c:
12259 media: fix default buffer size
12261 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12263 * gst/rtsp-server/rtsp-media-factory.c:
12264 * gst/rtsp-server/rtsp-media-factory.h:
12265 media-factory: add property to configure the buffer-size
12266 Add a property to configure the kernel UDP buffer size.
12268 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12270 * gst/rtsp-server/rtsp-media.c:
12271 * gst/rtsp-server/rtsp-media.h:
12272 media: add property to configure kernel buffer sizes
12273 Add a property to configure the kernel UDP buffer size.
12275 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12278 configure: set PYGOBJECT_REQ before using it
12279 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12281 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12283 * docs/Makefile.am:
12284 docs: recursive into sub-directories on 'make upload'
12286 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12288 * docs/libs/gst-rtsp-server-docs.sgml:
12289 * docs/version.entities.in:
12290 docs: mention full version these docs are for, not just major-minor
12292 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12295 back to development
12297 === release 0.10.8 ===
12299 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12304 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12306 * gst/rtsp-server/rtsp-server.c:
12307 rtsp-server: clarify docs a little
12309 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12311 * gst/rtsp-server/rtsp-media.c:
12312 media: init debug category before starting thread
12314 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12316 * gst/rtsp-server/rtsp-auth.c:
12317 auth: add realm to make it more spec compliant
12319 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12321 * gst/rtsp-server/rtsp-server.c:
12322 * gst/rtsp-server/rtsp-server.h:
12323 server: add locking
12325 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12327 * examples/test-video.c:
12328 example: improve example docs a little
12330 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12332 * gst/rtsp-server/rtsp-server.c:
12333 server: ensure the watch has a ref to the server
12335 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12337 * gst/rtsp-server/rtsp-server.c:
12338 server: simpify channel function
12340 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12342 * gst/rtsp-server/rtsp-server.c:
12343 * gst/rtsp-server/rtsp-server.h:
12344 server: simplify management of channel and source
12345 We don't need to keep around the channel and source objects. Let the mainloop
12346 and the source manage the source and channel respectively.
12348 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12354 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12356 * tests/.gitignore:
12357 * tests/Makefile.am:
12358 * tests/test-cleanup.c:
12359 tests: add tests directory and cleanup test
12361 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12363 * gst/rtsp-server/rtsp-media-factory-uri.c:
12364 * gst/rtsp-server/rtsp-media-factory.c:
12365 * gst/rtsp-server/rtsp-media-mapping.c:
12366 * gst/rtsp-server/rtsp-media.c:
12367 * gst/rtsp-server/rtsp-session-pool.c:
12368 * gst/rtsp-server/rtsp-session.c:
12369 server: improve debugging in various objects
12371 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12373 * gst/rtsp-server/rtsp-server.c:
12374 server: chain up to the parent finalize
12376 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12378 * bindings/python/rtspserver-types.defs:
12379 * bindings/python/rtspserver.defs:
12380 * bindings/python/rtspserver.override:
12381 * bindings/python/test.py:
12382 gst-rtsp-server: update python bindings
12384 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12386 * gst/rtsp-server/rtsp-client.c:
12387 client: use the response from the clientstate
12388 Create the response object only once and store in the client state.
12389 Make all methods use the state response,
12391 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12393 * gst/rtsp-server/rtsp-server.c:
12394 server: use signal to keep track of clients
12395 Keep track of all the clients that the server creates and remove them when they
12396 fire the 'closed' signal.
12398 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12400 * gst/rtsp-server/rtsp-client.c:
12401 * gst/rtsp-server/rtsp-client.h:
12402 client: emit signal when closing
12404 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12406 * examples/.gitignore:
12407 * examples/Makefile.am:
12408 * examples/test-auth.c:
12409 * examples/test-video.c:
12410 * gst/rtsp-server/rtsp-auth.c:
12411 * gst/rtsp-server/rtsp-auth.h:
12412 * gst/rtsp-server/rtsp-client.c:
12413 * gst/rtsp-server/rtsp-media-factory.c:
12414 * gst/rtsp-server/rtsp-media.c:
12415 * gst/rtsp-server/rtsp-media.h:
12416 * gst/rtsp-server/rtsp-session-pool.h:
12417 * gst/rtsp-server/rtsp-session.h:
12418 media: enable per factory authorisations
12419 Allow for adding a GstRTSPAuth on the factory and media level and check
12420 permissions when accessing the factory.
12421 Add hints to the auth methods for future more fine grained authorisation.
12422 Add example application for per factory authentication.
12424 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12426 * gst/rtsp-server/rtsp-auth.c:
12427 * gst/rtsp-server/rtsp-auth.h:
12428 * gst/rtsp-server/rtsp-client.c:
12429 * gst/rtsp-server/rtsp-client.h:
12430 * gst/rtsp-server/rtsp-params.c:
12431 * gst/rtsp-server/rtsp-params.h:
12432 rtsp-server: Pass ClientState structure arround
12433 Pass the collected information for the ongoing request in a GstRTSPClientState
12434 structure that we can then pass around to simplify the method arguments. This
12435 will also be handy when we implement logging functionality.
12437 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12439 * gst/rtsp-server/rtsp-media-factory.c:
12440 * gst/rtsp-server/rtsp-media-factory.h:
12441 media-factory: add methods to configure authorisation
12443 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12445 * gst/rtsp-server/rtsp-client.c:
12446 client: unref auth in finalize
12448 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12450 * gst/rtsp-server/rtsp-server.c:
12451 server: unref auth in finalize
12453 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12455 * docs/libs/gst-rtsp-server-docs.sgml:
12456 * docs/libs/gst-rtsp-server-sections.txt:
12457 * docs/libs/gst-rtsp-server.types:
12458 docs: add more docs
12460 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12462 * gst/rtsp-server/rtsp-server.c:
12463 * gst/rtsp-server/rtsp-server.h:
12464 server: separate create and accept
12465 Create separate create and accept methods so that subclasses can create custom
12467 Configure the server in the client object and prepare for keeping track of
12470 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12472 * gst/rtsp-server/rtsp-client.c:
12473 * gst/rtsp-server/rtsp-client.h:
12474 client: add support for setting the server.
12475 Add support for keeping a ref to the server that started this client
12478 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12480 * gst/rtsp-server/rtsp-auth.c:
12481 auth: fix memleak and add some docs
12482 Fix a memleak of the basic auth token.
12483 Add docs for the helper function
12485 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12487 * gst/rtsp-server/rtsp-auth.c:
12488 * gst/rtsp-server/rtsp-auth.h:
12489 * gst/rtsp-server/rtsp-client.c:
12490 client: delegate setup of auth to the manager
12491 Delegate the configuration of the authentication tokens to the manager object
12494 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12496 * examples/test-video.c:
12497 * gst/rtsp-server/Makefile.am:
12498 * gst/rtsp-server/rtsp-auth.c:
12499 * gst/rtsp-server/rtsp-auth.h:
12500 * gst/rtsp-server/rtsp-client.c:
12501 * gst/rtsp-server/rtsp-client.h:
12502 * gst/rtsp-server/rtsp-server.c:
12503 * gst/rtsp-server/rtsp-server.h:
12504 auth: add authentication object
12505 Add an object that can check the authorization of requests.
12506 Implement basic authentication.
12507 Add example authentication to test-video
12509 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12511 * gst/rtsp-server/rtsp-server.c:
12512 * gst/rtsp-server/rtsp-server.h:
12513 server: move includes back
12514 the includes are needed for sockaddr_in.
12516 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12518 * gst/rtsp-server/rtsp-client.c:
12519 * gst/rtsp-server/rtsp-client.h:
12520 * gst/rtsp-server/rtsp-server.c:
12521 * gst/rtsp-server/rtsp-server.h:
12522 rtsp: move network includes where they are needed
12524 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12526 * gst/rtsp-server/rtsp-media.h:
12527 rtsp-media.h: Minor corrections in comments.
12530 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12533 Automatic update of common submodule
12534 From e572c87 to f94d739
12536 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12540 * docs/libs/.gitignore:
12541 * examples/.gitignore:
12542 * gst/rtsp-server/.gitignore:
12545 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12547 * docs/libs/Makefile.am:
12548 docs: We don't build ps/pdf for API reference docs
12550 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12553 Automatic update of common submodule
12554 From ccbaa85 to e572c87
12556 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12559 Automatic update of common submodule
12560 From 46445ad to ccbaa85
12562 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12564 * gst/rtsp-server/Makefile.am:
12565 * gst/rtsp-server/rtsp-funnel.c:
12566 * gst/rtsp-server/rtsp-funnel.h:
12567 * gst/rtsp-server/rtsp-media.c:
12568 funnel: rename fsfunnel to rtspfunnel
12569 Rename the funnel to avoid conflicts with the farsight one.
12571 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12573 * gst/rtsp-server/Makefile.am:
12574 * gst/rtsp-server/fs-funnel.c:
12575 * gst/rtsp-server/fs-funnel.h:
12576 * gst/rtsp-server/rtsp-media.c:
12577 rtsp-media: add and use fsfunnel
12578 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12579 select-all property that we need.
12581 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12583 * gst/rtsp-server/Makefile.am:
12584 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12585 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12586 for the g-ir-compiler, rather than just assuming the env var has
12589 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12596 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12598 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12601 * gst/rtsp-server/Makefile.am:
12602 gobject-introspection: fix g-i build for uninstalled setup
12603 Requires gst-plugins-base git (> 0.10.31.2).
12605 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12607 * examples/test-uri.c:
12608 examples: add some more options and comments
12610 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12612 * gst/rtsp-server/rtsp-media-factory-uri.c:
12613 factory-uri: use right property type
12615 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12617 * gst/rtsp-server/rtsp-media-factory-uri.c:
12618 factory-uri: attempt to configure buffer-lists
12619 Attempt to configure buffer lists in the payloader for improved performance.
12621 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12623 * gst/rtsp-server/rtsp-media.c:
12624 media: attempt to configure bigger UDP buffers
12625 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12626 send buffers with high bitrate streams.
12628 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12630 * gst/rtsp-server/rtsp-client.c:
12631 client: use the socket length from getsockname
12632 Use the length returned by getsockname to perform the getnameinfo call because
12633 the size can depend on the socket type and platform.
12636 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12638 * docs/libs/gst-rtsp-server-docs.sgml:
12639 * docs/libs/gst-rtsp-server-sections.txt:
12640 docs: add uri factory to the docs
12642 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12644 * gst/rtsp-server/rtsp-client.c:
12645 * gst/rtsp-server/rtsp-media.h:
12648 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12650 * gst/rtsp-server/rtsp-client.c:
12651 * gst/rtsp-server/rtsp-media.c:
12652 * gst/rtsp-server/rtsp-media.h:
12653 * gst/rtsp-server/rtsp-session.c:
12654 * gst/rtsp-server/rtsp-session.h:
12655 rtsp-server: add support for buffer lists
12656 Add support for sending bufferlists received from appsink.
12659 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12661 * gst/rtsp-server/rtsp-client.c:
12662 * gst/rtsp-server/rtsp-media.c:
12663 * gst/rtsp-server/rtsp-media.h:
12664 * gst/rtsp-server/rtsp-sdp.c:
12665 media: make method to retrieve the play range
12666 Make a method to retrieve the playback range so that we can conditionally create
12667 a different range for the SDP and the PLAY requests.
12669 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12671 * gst/rtsp-server/rtsp-media.c:
12672 * gst/rtsp-server/rtsp-media.h:
12673 media: add signal to notify of state changes
12675 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12677 * gst/rtsp-server/rtsp-client.h:
12678 client: cleanup headers
12680 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12682 * gst/rtsp-server/rtsp-client.c:
12685 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12687 * gst/rtsp-server/rtsp-media-factory-uri.c:
12688 * gst/rtsp-server/rtsp-media-factory-uri.h:
12689 factory-uri: add support for gstpay
12690 Add an option to prefer gstpay over decoder + raw payloader.
12692 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12694 * gst/rtsp-server/rtsp-media-factory-uri.c:
12695 * gst/rtsp-server/rtsp-media-factory-uri.h:
12696 factory-uri: rework the autoplugger.
12697 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12700 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12702 * gst/rtsp-server/rtsp-media-factory-uri.c:
12703 factory-uri: use better factory filter
12704 Make better payloader filter based on autoplug rank and RTP use case.
12706 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12709 Automatic update of common submodule
12710 From 169462a to 46445ad
12712 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12714 * gst/rtsp-server/rtsp-server.c:
12715 server: set SO_REUSEADDR before bind
12716 Set the SO_REUSEADDR _before_ bind() to make it actually work.
12718 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12720 * gst/rtsp-server/rtsp-media.c:
12721 * gst/rtsp-server/rtsp-media.h:
12722 media: emit prepared signal when prepared
12723 Make a 'prepared' signal and emit it when we successfully prepared the element.
12724 This signal can be used to configure the media object after it has been prepared
12727 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
12730 Automatic update of common submodule
12731 From 011bcc8 to 169462a
12733 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
12735 python an optional dependency
12736 * configure.ac: Move up valgrind and g-i checks. Make the python
12737 dependency optional, as it was before.
12739 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12741 Merge branch 'master' into 0.11
12746 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12748 * gst/rtsp-server/rtsp-media.c:
12749 media: update range when active clients changed
12750 When we changed the number of active clients, update the current range
12751 information because we want the second client connecting to a shared resource
12752 continue from where the stream currently.
12754 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12756 * gst/rtsp-server/rtsp-media-factory-uri.c:
12757 * gst/rtsp-server/rtsp-media-factory-uri.h:
12758 factory-uri: add colorspace and fix pt
12759 Rework the way we pass data to the autoplugger.
12760 When we have raw caps, plug a converter element to make pluggin to raw
12761 payloaders more successful.
12762 Make sure all dynamically plugged payloaders have a unique payload types.
12764 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12766 * examples/Makefile.am:
12767 * examples/test-uri.c:
12768 example: add example of the uri factory
12770 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12772 * gst/rtsp-server/Makefile.am:
12773 * gst/rtsp-server/rtsp-media-factory-uri.c:
12774 * gst/rtsp-server/rtsp-media-factory-uri.h:
12775 * gst/rtsp-server/rtsp-server.h:
12776 factory-uri: add a factory to stream any URI
12777 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
12780 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12782 * gst/rtsp-server/rtsp-media.c:
12783 * gst/rtsp-server/rtsp-media.h:
12784 media: ignore spurious ASYNC_DONE messages
12785 When we are dynamically adding pads, the addition of the udpsrc elements will
12786 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
12787 the real ASYNC_DONE when everything is prerolled.
12789 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12791 * gst/rtsp-server/rtsp-media-factory.c:
12792 * gst/rtsp-server/rtsp-media-factory.h:
12793 media-factory: make lock macro
12795 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
12797 * gst/rtsp-server/rtsp-client.c:
12798 rtsp-server: Remove unused variable and dead assignment
12800 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
12802 * examples/test-launch.c:
12803 * examples/test-mp4.c:
12804 * examples/test-ogg.c:
12805 * examples/test-readme.c:
12806 * examples/test-sdp.c:
12807 * examples/test-video.c:
12808 examples: Run gst-indent
12810 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
12812 * gst/rtsp-server/rtsp-client.c:
12813 * gst/rtsp-server/rtsp-media-factory.c:
12814 * gst/rtsp-server/rtsp-media-mapping.c:
12815 * gst/rtsp-server/rtsp-media.c:
12816 * gst/rtsp-server/rtsp-params.c:
12817 * gst/rtsp-server/rtsp-sdp.c:
12818 * gst/rtsp-server/rtsp-server.c:
12819 * gst/rtsp-server/rtsp-session-pool.c:
12820 * gst/rtsp-server/rtsp-session.c:
12821 rtsp-server: Run gst-indent
12822 Since it wasn't using the upstream common previously, there was no
12823 indentation check before commiting.
12825 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
12827 * gst/rtsp-server/rtsp-media-mapping.h:
12828 * gst/rtsp-server/rtsp-media.c:
12829 * gst/rtsp-server/rtsp-media.h:
12830 * gst/rtsp-server/rtsp-sdp.c:
12831 * gst/rtsp-server/rtsp-session-pool.h:
12832 * gst/rtsp-server/rtsp-session.c:
12833 * gst/rtsp-server/rtsp-session.h:
12834 rtsp-server: Some more doc fixups
12836 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12839 Makefile: Add cruft-cleaning support
12841 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12845 * docs/Makefile.am:
12846 * docs/libs/Makefile.am:
12847 * docs/libs/gst-rtsp-server-docs.sgml:
12848 * docs/libs/gst-rtsp-server-sections.txt:
12849 * docs/libs/gst-rtsp-server.types:
12850 * docs/version.entities.in:
12851 docs: Add gtk-doc build system
12853 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12855 * gst/rtsp-server/Makefile.am:
12856 Makefile.am: Use standard GIR make behaviour
12858 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12862 autogen/configure: Bring more in sync to standard gst module behaviour
12864 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12866 * gst/rtsp-server/rtsp-media.c:
12867 media: warn and fail when gstrtpbin is not found
12869 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12872 configure: open 0.11 branch
12874 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
12878 Add common submodule
12880 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
12882 * common/ChangeLog:
12883 * common/Makefile.am:
12884 * common/c-to-xml.py:
12885 * common/check.mak:
12886 * common/coverage/coverage-report-entry.pl:
12887 * common/coverage/coverage-report.pl:
12888 * common/coverage/coverage-report.xsl:
12889 * common/coverage/lcov.mak:
12890 * common/gettext.patch:
12891 * common/glib-gen.mak:
12892 * common/gst-autogen.sh:
12893 * common/gst-xmlinspect.py:
12895 * common/gstdoc-scangobj:
12896 * common/gtk-doc-plugins.mak:
12897 * common/gtk-doc.mak:
12898 * common/m4/.gitignore:
12899 * common/m4/Makefile.am:
12900 * common/m4/README:
12901 * common/m4/as-ac-expand.m4:
12902 * common/m4/as-auto-alt.m4:
12903 * common/m4/as-compiler-flag.m4:
12904 * common/m4/as-compiler.m4:
12905 * common/m4/as-docbook.m4:
12906 * common/m4/as-libtool-tags.m4:
12907 * common/m4/as-libtool.m4:
12908 * common/m4/as-python.m4:
12909 * common/m4/as-scrub-include.m4:
12910 * common/m4/as-version.m4:
12911 * common/m4/ax_create_stdint_h.m4:
12912 * common/m4/check.m4:
12913 * common/m4/glib-gettext.m4:
12914 * common/m4/gst-arch.m4:
12915 * common/m4/gst-args.m4:
12916 * common/m4/gst-check.m4:
12917 * common/m4/gst-debuginfo.m4:
12918 * common/m4/gst-default.m4:
12919 * common/m4/gst-doc.m4:
12920 * common/m4/gst-error.m4:
12921 * common/m4/gst-feature.m4:
12922 * common/m4/gst-function.m4:
12923 * common/m4/gst-gettext.m4:
12924 * common/m4/gst-glib2.m4:
12925 * common/m4/gst-libxml2.m4:
12926 * common/m4/gst-plugindir.m4:
12927 * common/m4/gst-valgrind.m4:
12928 * common/m4/gtk-doc.m4:
12929 * common/m4/introspection.m4:
12930 * common/m4/pkg.m4:
12931 * common/mangle-tmpl.py:
12932 * common/plugins.xsl:
12934 * common/release.mak:
12935 * common/scangobj-merge.py:
12936 * common/upload.mak:
12937 common: Remove static version
12939 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
12941 * common/m4/introspection.m4:
12942 Update introspection.m4 to match usage
12944 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12948 Remove old stuff from the README
12950 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12953 back to development
12955 === release 0.10.7 ===
12957 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12962 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12964 * examples/test-ogg.c:
12965 test-ogg: remove parsers
12966 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
12967 buffers with timestamps. Using the parsers also seems to break things.
12969 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12971 * bindings/vala/gst-rtsp-server-0.10.vapi:
12972 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12973 Updated Vala bindings
12975 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12977 * common/m4/introspection.m4:
12979 * gst/rtsp-server/Makefile.am:
12980 Added initial gobject-introspection support
12982 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12984 * gst/rtsp-server/rtsp-media-factory.c:
12985 media-factory: don't use host for shared hash key
12986 When we generate the key to share made between connections, don't include the
12987 host used to connect so that we can share media even if between clients that
12988 connected with localhost and ones with the ip address.
12990 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12992 * bindings/vala/Makefile.am:
12993 build: fix distcheck
12995 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12997 * bindings/vala/gst-rtsp-server-0.10.vapi:
12998 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12999 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13000 Update Vala bindings
13002 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13004 * bindings/vala/Makefile.am:
13006 Fix configure checks and installation location for Vala bindings
13009 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13012 back to development
13014 === release 0.10.6 ===
13016 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13019 configure: release 0.10.6
13021 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13023 * gst/rtsp-server/rtsp-media.c:
13024 media: help the compiler a little
13026 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13028 * gst/rtsp-server/rtsp-media.c:
13029 * gst/rtsp-server/rtsp-media.h:
13030 * gst/rtsp-server/rtsp-session.c:
13031 media: cleanup media transport before freeing
13032 Cleanup the media transport data before freeing. In particular, remove the qdata
13033 from the rtpsource object.
13035 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13037 * gst/rtsp-server/rtsp-media-factory.c:
13038 * gst/rtsp-server/rtsp-media-factory.h:
13039 * gst/rtsp-server/rtsp-media.c:
13040 * gst/rtsp-server/rtsp-media.h:
13041 media-factory: add eos-shutdown property
13042 Add an eos-shutdown property that will send an EOS to the pipeline before
13043 shutting it down. This allows for nice cleanup in case of a muxer.
13046 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13048 * gst/rtsp-server/rtsp-media.c:
13049 * gst/rtsp-server/rtsp-media.h:
13050 media: use multiudpsink send-duplicates when we can
13051 If we have a new enough multiudpsink with the send-duplicates property, use this
13052 instead of doing our own filtering. Our custom filtering code should eventually
13053 be removed when we can depend on a released -good.
13055 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13057 * gst/rtsp-server/rtsp-media.c:
13058 media: don't leak destinations
13059 Refactor and cleanup the destinations array when the stream is destroyed.
13061 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13063 * gst/rtsp-server/rtsp-media.c:
13064 * gst/rtsp-server/rtsp-media.h:
13065 media: don't add udp addresses multiple times
13066 Keep track of the udp addresses we added to udpsink and never add the same udp
13067 destination twice. This avoids duplicate packets when using multicast.
13069 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13071 * gst/rtsp-server/rtsp-server.c:
13072 server: disable use of SO_LINGER
13073 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13074 server close()s the connection.
13076 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13078 * gst/rtsp-server/rtsp-server.c:
13079 server: use 5 second linger period in SO_LINGER
13080 Wait 5 seconds before clearing the send buffers and reseting the connection with
13081 the client when we do a close. This should be enough time to get the message to
13085 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13087 * gst/rtsp-server/rtsp-server.c:
13088 server: use SO_LINGER
13089 SO_LINGER on the socket will make sure that any pending data on the socket is
13090 flushed ASAP and that the socket connection is reset. This makes sure that the
13091 socket can be reused immediately.
13094 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13097 README: add blurb about shared media factories
13099 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13101 * gst/rtsp-server/rtsp-media.c:
13102 Add stdlib.h for atoi()
13104 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13106 * bindings/python/Makefile.am:
13107 * bindings/vala/Makefile.am:
13108 build: distcheck fixes
13109 Fix 'make distcheck', somewhat (it still fails because it tries to
13110 install files into /usr/share/vala/vapi/ irrespective of the
13111 configured prefix).
13113 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13116 configure: bump core/base requirements to released version
13117 Makes things less confusing for people.
13119 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13122 configure: fail if GStreamer core/base requirements are not met
13124 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13126 * gst/rtsp-server/rtsp-client.c:
13127 client: improve client cleanups
13128 Make sure the session does not timeout when using TCP. We need to do this
13129 because quicktime player does not send RTCP for some reason in tunneled
13131 Refactor some cleanup code.
13134 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13136 * gst/rtsp-server/rtsp-session.c:
13137 * gst/rtsp-server/rtsp-session.h:
13138 session: add support for prevent session timeouts
13139 Add an atomix counter to prevent session timeouts when we are, for example,
13140 streaming over TCP.
13142 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13144 * gst/rtsp-server/rtsp-client.c:
13145 client: fix unlink on session timeouts
13146 When our session times out, make sure we unlink all streams in this
13148 Remove the tunnelid when closing the connection.
13150 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13152 * gst/rtsp-server/rtsp-session.c:
13153 session: small cleanups
13155 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13157 * gst/rtsp-server/rtsp-client.c:
13158 client: handle lost_tunnel callbacks
13159 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13160 hashtable so that we can reuse it for when the client reopens the POST
13162 Close the connection after a TEARDOWN.
13163 Make sure or watchid is cleared when the watch is removed.
13166 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13168 * gst/rtsp-server/rtsp-client.c:
13169 * gst/rtsp-server/rtsp-media.c:
13170 * gst/rtsp-server/rtsp-sdp.c:
13171 rtsp-server: add more support for multicast
13173 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13176 * gst/rtsp-server/rtsp-media.c:
13177 * gst/rtsp-server/rtsp-media.h:
13178 media: allow configuration of allowed lower transport
13180 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13182 * gst/rtsp-server/rtsp-client.h:
13183 * gst/rtsp-server/rtsp-media.c:
13184 * gst/rtsp-server/rtsp-media.h:
13185 * gst/rtsp-server/rtsp-sdp.c:
13186 * gst/rtsp-server/rtsp-sdp.h:
13187 * gst/rtsp-server/rtsp-server.c:
13188 rtsp: keep track of server ip and ipv6
13189 Keep track of how the client connected to the server and setup the udp ports
13190 with the same protocol.
13191 Copy the server ip address in the SDP so that clients can send RTCP back to
13194 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13196 * gst/rtsp-server/rtsp-session.c:
13199 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13201 * gst/rtsp-server/rtsp-client.c:
13202 client: use right size for malloc
13204 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13206 * gst/rtsp-server/rtsp-server.c:
13207 server: comment ipv6 server listening address
13209 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13211 * gst/rtsp-server/rtsp-media.c:
13212 media: allow for ipv6 sockets
13214 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13216 * gst/rtsp-server/rtsp-server.c:
13217 * gst/rtsp-server/rtsp-server.h:
13218 server: rework server part
13219 Allow setting a bind address, make sure we can deal with ipv6.
13220 Remove the port property and change with the service property.
13222 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13224 * gst/rtsp-server/rtsp-media.h:
13225 media: update comments a little
13227 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13229 * gst/rtsp-server/rtsp-client.c:
13230 client: make content-base better
13231 Use the URI formatting functions to make a content-base. Also make sure that
13232 there is a trailing / at the end.
13234 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13236 * gst/rtsp-server/rtsp-client.c:
13237 client: guard against invalid paths
13239 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13241 * examples/test-video.c:
13242 test: catch server bind errors
13244 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13246 * gst/rtsp-server/rtsp-media.c:
13247 rtspmedia: emit "unprepared" if _prepare fails.
13248 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13249 media object is removed from its factory's cache.
13251 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13253 * gst/rtsp-server/rtsp-media.c:
13254 media: collect media position when seek completes
13256 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13258 * gst/rtsp-server/rtsp-client.c:
13259 client: call unlink_streams in client finalize
13262 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13264 * gst/rtsp-server/rtsp-media.c:
13265 media: limit the time to wait to something huge
13266 Avoid waiting forever but limit the timeout to 20 seconds.
13268 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13270 * gst/rtsp-server/rtsp-sdp.c:
13271 sdp: reindent and check for prepared status
13273 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13275 * gst/rtsp-server/rtsp-media.c:
13276 * gst/rtsp-server/rtsp-media.h:
13277 * gst/rtsp-server/rtsp-session.c:
13278 media: avoid doing _get_state() for state changes
13279 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13280 until the media is prerolled or in error. This avoids doing a blocking call of
13281 gst_element_get_state() that can cause lockups when there is an error.
13284 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13286 * gst/rtsp-server/rtsp-media.c:
13289 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13291 * gst/rtsp-server/rtsp-media-factory.c:
13292 media-factory: better error handling
13293 Improve the error handling a bit.
13295 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13297 * gst/rtsp-server/rtsp-client.c:
13298 client: rework transport parsing
13299 Rework the transport parsing code so that we can ignore transports we don't
13300 support instead of just picking the first one we can parse.
13301 Configure a (for now hardcoded) destination for multicast transports.
13303 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13305 * gst/rtsp-server/rtsp-media.c:
13306 media: set multicast sink parameters
13307 Disable loop and automatic multicast join on the udpsink elements.
13308 Add some more debug info.
13309 Reset some state variables in the right place.
13310 Use the right port numbers for multicast.
13312 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13314 * gst/rtsp-server/rtsp-session.c:
13315 session: handle transport setup correctly
13316 Handle UDP, MCAST and TCP transport negotiation more correctly.
13317 Store the server session SSRC in the transport.
13319 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13321 * gst/rtsp-server/rtsp-client.c:
13322 rtsp-client: implement error_full
13323 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13326 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13329 * gst/rtsp-server/rtsp-client.c:
13330 * gst/rtsp-server/rtsp-server.c:
13331 docs: update docs and comments
13333 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13335 * gst/rtsp-server/rtsp-sdp.c:
13336 sdp: make server work better when behind a proxy
13338 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13340 * gst/rtsp-server/rtsp-client.c:
13341 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13343 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13345 * gst/rtsp-server/rtsp-client.c:
13346 * gst/rtsp-server/rtsp-media-factory.c:
13347 * gst/rtsp-server/rtsp-media-mapping.c:
13348 * gst/rtsp-server/rtsp-media.c:
13349 * gst/rtsp-server/rtsp-server.c:
13350 * gst/rtsp-server/rtsp-session-pool.c:
13351 * gst/rtsp-server/rtsp-session.c:
13352 Use GStreamer's debugging subsystem
13354 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13356 * gst/rtsp-server/rtsp-media-factory.c:
13357 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13359 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13362 back to development
13364 === release 0.10.5 ===
13366 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13371 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13374 configure: bump required versions
13376 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13378 * gst/rtsp-server/rtsp-client.c:
13379 client: call weak-unref on client->sessions from finalize
13382 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13384 * gst/rtsp-server/rtsp-media.c:
13385 media: Fixed crasher where caps got unref'ed too often
13387 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13390 * pkgconfig/.gitignore:
13391 * pkgconfig/Makefile.am:
13392 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13393 Added pkg-config file to use gst-rtsp-server uninstalled
13395 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13397 * gst/rtsp-server/rtsp-media.c:
13398 media: add some docs
13400 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13402 * gst/rtsp-server/rtsp-client.c:
13403 rtsp: Use gst_rtsp_watch_send_message().
13404 Use gst_rtsp_watch_send_message() since the old API which used
13405 gst_rtsp_watch_queue_message() has been deprecated.
13407 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13410 back to development
13412 === release 0.10.4 ===
13414 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13419 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13421 * gst/rtsp-server/rtsp-client.c:
13422 * gst/rtsp-server/rtsp-session.c:
13423 * gst/rtsp-server/rtsp-session.h:
13424 rtsp: allocate channels in TCP mode
13425 When the client does not provide us with channels in TCP mode, allocate channels
13428 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13430 * gst/rtsp-server/rtsp-client.c:
13431 client: don't crash when tunnelid is missing
13432 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13433 don't crash but return an error response to the client.
13436 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13438 * bindings/vala/gst-rtsp-server-0.10.vapi:
13439 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13440 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13441 bindings: update vala bindings with new method
13443 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13445 * gst/rtsp-server/rtsp-session-pool.c:
13446 * gst/rtsp-server/rtsp-session-pool.h:
13447 sessionpool: add function to filter sessions
13448 Add generic function to retrieve/remove sessions.
13450 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13453 configure: bump core/base requirements to release
13455 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13457 * gst/rtsp-server/rtsp-media.c:
13458 media: fix indentation
13460 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13462 * gst/rtsp-server/rtsp-media.c:
13463 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13465 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13467 * gst/rtsp-server/rtsp-media.c:
13468 set state and remove elements of media in for loop
13470 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13472 * bindings/vala/gst-rtsp-server-0.10.vapi:
13473 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13474 Added gst_rtsp_media_remove_elements function to Vala bindings
13476 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13478 * gst/rtsp-server/rtsp-media.c:
13479 * gst/rtsp-server/rtsp-media.h:
13480 Added gst_rtsp_media_remove_elements function
13482 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13484 * gst/rtsp-server/rtsp-media.c:
13485 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13487 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13489 * bindings/vala/gst-rtsp-server-0.10.vapi:
13490 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13491 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13492 Updated Vala bindings
13494 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13496 * gst/rtsp-server/rtsp-media.c:
13497 * gst/rtsp-server/rtsp-media.h:
13498 Added vmethod unprepare to GstRTSPMedia
13499 The default implementation sets the state of the pipeline to GST_STATE_NULL
13501 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13503 * gst/rtsp-server/rtsp-media-factory.c:
13504 * gst/rtsp-server/rtsp-media-factory.h:
13505 Made collect_streams function public
13507 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13509 * gst/rtsp-server/rtsp-media-factory.c:
13510 * gst/rtsp-server/rtsp-media-factory.h:
13511 * gst/rtsp-server/rtsp-media.c:
13512 Added vmethod create_pipeline to GstRTSPMediaFactory
13513 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13515 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13517 * gst/rtsp-server/rtsp-client.c:
13518 client: use g_source_destroy()
13519 We need to use g_source_destroy() because we might have added the source to a
13520 different main context than the default one.
13522 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13524 * gst/rtsp-server/Makefile.am:
13525 * gst/rtsp-server/rtsp-client.c:
13526 * gst/rtsp-server/rtsp-params.c:
13527 * gst/rtsp-server/rtsp-params.h:
13528 rtsp: prepare for handling GET/SET_PARAMETER
13529 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13531 Fix return codes of handlers.
13533 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13535 * gst/rtsp-server/rtsp-media.c:
13536 media: don't leak session pads
13538 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13540 * gst/rtsp-server/rtsp-media.c:
13541 media: clean up the messages a bit
13543 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13545 * gst/rtsp-server/rtsp-sdp.c:
13546 sdp: warn and skip streams without media
13548 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13550 * bindings/vala/gst-rtsp-server-0.10.vapi:
13551 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13552 vala: Fixed typo in header file of RTSPMediaStream
13554 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13556 * gst/rtsp-server/rtsp-media.c:
13558 Fix a debug message
13559 Make dumping RTCP stats configurable
13561 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13563 * gst/rtsp-server/rtsp-media.c:
13564 media: be less verbose and leak less
13566 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13568 * gst/rtsp-server/rtsp-media.c:
13569 media: don't leak the destination address
13571 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13573 * gst/rtsp-server/rtsp-client.c:
13574 * gst/rtsp-server/rtsp-media.c:
13575 * gst/rtsp-server/rtsp-media.h:
13576 * gst/rtsp-server/rtsp-session.c:
13577 * gst/rtsp-server/rtsp-session.h:
13578 rtsp: use RTCP to keep the session alive
13579 Use the RTCP rtcp-from stats field to find the associated session and use this
13580 to keep the session alive.
13582 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13584 * gst/rtsp-server/rtsp-session.c:
13585 session: add 5sec to the real session timeout
13586 Allow the session to live 5sec longer before really timing out. This should give
13587 clients some extra time to keep the session active.
13589 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13591 * gst/rtsp-server/rtsp-client.c:
13592 client: replay OK to GET/SET_PARAMETER
13593 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13594 so that we return OK for those requests.
13596 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13598 * gst/rtsp-server/rtsp-media.c:
13599 * gst/rtsp-server/rtsp-media.h:
13600 media: keep track of active transports
13601 Keep track of which transport is active to avoid closing the connection too
13603 Remove the destination transport also when going to NULL.
13604 Print some stats about the SDES and other RTCP messages we receive from the
13607 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13609 * examples/.gitignore:
13610 * examples/Makefile.am:
13611 * examples/test-sdp.c:
13612 example: add SDP relay example
13614 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13616 * gst/rtsp-server/rtsp-media.c:
13617 media: also count active TCP connections
13619 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13621 * gst/rtsp-server/rtsp-media-factory.c:
13622 * gst/rtsp-server/rtsp-media.c:
13623 * gst/rtsp-server/rtsp-media.h:
13624 rtsp: add support for dynamic elements
13625 Add support for dynamic elements.
13626 Don't set live pipelines back to paused.
13628 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13630 * gst/rtsp-server/rtsp-sdp.c:
13631 sdp: don't add encoding name when absent in caps
13633 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13635 * gst/rtsp-server/rtsp-client.c:
13636 client: warn when we can't do RTP-Info
13638 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13640 * gst/rtsp-server/rtsp-media-factory.c:
13641 factory: factor out the stream construction
13643 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13645 * gst/rtsp-server/rtsp-client.c:
13646 client: only add RTP-Info when we have the info
13647 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13650 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13653 back to development
13655 === release 0.10.3 ===
13657 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13661 - Fixes a bug where it put the wrong verion in pkgconfig
13662 - Link RTP and RTCP sources
13664 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13666 * gst/rtsp-server/rtsp-media.c:
13667 * gst/rtsp-server/rtsp-media.h:
13668 media: link the RTP udpsrc to the session manager
13669 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13670 shut down when the client sends a packet to open firewalls.
13672 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13674 * pkgconfig/gst-rtsp-server.pc.in:
13675 Don't use hard-coded version number in pkg-config file
13677 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13680 back to development
13682 === release 0.10.2 ===
13684 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13689 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13692 * common/m4/.gitignore:
13693 * examples/.gitignore:
13694 * pkgconfig/.gitignore:
13695 add some .gitignore files
13697 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13699 * gst/rtsp-server/rtsp-media.c:
13700 media: seek to key frames
13702 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13704 * gst/rtsp-server/rtsp-media.c:
13705 media: emit the unprepared signal by id
13706 Emit the unprepared signal by id instead of name and set the media as
13709 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13711 * gst/rtsp-server/rtsp-media.c:
13712 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13714 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13716 * gst/rtsp-server/rtsp-server.c:
13717 Added finalize function to GstRTPSPServer to unref session pool and media mapping
13719 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13721 * bindings/vala/gst-rtsp-server-0.10.vapi:
13722 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13723 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13724 Updated vala bindings
13726 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13728 * gst/rtsp-server/Makefile.am:
13729 * gst/rtsp-server/rtsp-client.c:
13730 * gst/rtsp-server/rtsp-media.c:
13731 server: use appsink and appsrc with the API
13732 Use the appsink/appsrc API instead of the signals for higher
13735 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13737 * examples/test-ogg.c:
13738 tests: set the payload type correctly
13740 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13742 * gst/rtsp-server/rtsp-media-factory.c:
13743 factory: connect to the unprepare signal
13744 Connect to the unprepare signal for non-reusable media so that we can remove
13745 them from the cache.
13747 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13749 * gst/rtsp-server/rtsp-media.c:
13750 * gst/rtsp-server/rtsp-media.h:
13751 media: add signal to notify of unprepare
13753 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13755 * gst/rtsp-server/rtsp-media.c:
13756 * gst/rtsp-server/rtsp-media.h:
13757 media: more work on making the media shared
13758 Add a reusable flag to medias, indicating that they can be reused after a state
13762 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13764 * examples/test-readme.c:
13765 examples: mark the example as shared for testing
13767 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13769 * gst/rtsp-server/rtsp-media.c:
13770 * gst/rtsp-server/rtsp-media.h:
13771 client: support shared media
13772 Always perform the state actions even if the target state of the pipeline is
13773 already correct, we still want to add/remove the transports when we are dealing
13775 Keep a counter of the number of active transports for a media so that we can use
13776 this to perform a state change when needed.
13777 Perform a state change of the pipeline only when the first transport was added
13778 or when there are no active transports.
13780 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13782 * gst/rtsp-server/rtsp-client.c:
13783 client: fix refcounting crasher
13784 Don't need to remove the weak refs in the finalize methods, they are already
13785 removed in the dispose.
13786 Don't register the callback with a DestroyNofity.
13788 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13790 * gst/rtsp-server/rtsp-client.c:
13791 Fix rtsp client refcount management in TCP mode.
13792 Don't unref a client ref we never had. Fixes an unref
13793 of an already-free client object after a client
13794 teardown request for me.
13796 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13798 * gst/rtsp-server/rtsp-session.c:
13799 docs: fix typo in API docs
13801 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13803 * gst/rtsp-server/rtsp-media.c:
13804 More seeking fixes.
13805 Keep the udp sources in playing even if we go to paused. unlock the sources when
13807 Add some more debug info.
13808 Only seek when we need to.
13809 Keep track of the position when we go to paused.
13811 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13813 * gst/rtsp-server/rtsp-client.c:
13814 * gst/rtsp-server/rtsp-media.c:
13815 * gst/rtsp-server/rtsp-media.h:
13816 Add beginnings of seeking.
13817 Parse the Range header and perform a seek on the pipeline for the requested
13818 position. It's disabled currently until I figure out what's going wrong.
13820 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13822 * gst/rtsp-server/rtsp-client.c:
13823 allow pause requests for now.
13826 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13828 * gst/rtsp-server/rtsp-client.c:
13829 Remove weak ref on the session in teardown
13830 We need to remove our weakref from the session when we do a teardown because
13831 else we close the TCP connection prematurely.
13833 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13835 * gst/rtsp-server/rtsp-client.c:
13836 * gst/rtsp-server/rtsp-client.h:
13837 * gst/rtsp-server/rtsp-session-pool.c:
13838 Do some more session cleanup
13839 Make session timeout kill the TCP connection that currently watches the
13841 Remove the client timeout property.
13843 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13845 * gst/rtsp-server/rtsp-client.c:
13846 * gst/rtsp-server/rtsp-client.h:
13847 * gst/rtsp-server/rtsp-media.c:
13848 * gst/rtsp-server/rtsp-media.h:
13849 * gst/rtsp-server/rtsp-server.c:
13850 * gst/rtsp-server/rtsp-session.c:
13851 * gst/rtsp-server/rtsp-session.h:
13853 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
13856 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13858 * examples/Makefile.am:
13859 * examples/test-launch.c:
13860 Add example server that takes launch lines
13861 Add an example server that streams any -launch line.
13863 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13865 * examples/test-readme.c:
13866 * gst/rtsp-server/rtsp-client.c:
13867 * gst/rtsp-server/rtsp-media.c:
13868 * gst/rtsp-server/rtsp-media.h:
13869 Add support for live streams
13870 Add support for live streams and ranges
13871 Start on handling TCP data transfer.
13873 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13875 * gst/rtsp-server/rtsp-media.c:
13876 Free the pipeline before other things
13879 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13881 * gst/rtsp-server/rtsp-client.c:
13882 Only free the pending tunnel if there is one
13885 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13887 * gst/rtsp-server/rtsp-client.c:
13888 * gst/rtsp-server/rtsp-client.h:
13889 * gst/rtsp-server/rtsp-media.c:
13890 rtsp-server: Add support for tunneling
13891 Add support for tunneling over HTTP.
13892 Use new connection methods to retrieve the url.
13893 Dispatch messages based on the message type instead of blindly
13894 assuming it's always a request.
13895 Keep track of the watch id so that we can remove it later.
13896 Set the media pipeline to NULL before unreffing the pipeline.
13898 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13900 * gst/rtsp-server/rtsp-client.c:
13901 * gst/rtsp-server/rtsp-client.h:
13902 Fix for channel -> watch rename in gstreamer
13903 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
13905 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13907 * gst/rtsp-server/rtsp-client.c:
13908 * gst/rtsp-server/rtsp-client.h:
13910 Use the async RTSP channels instead of spawning a new thread for each client.
13911 If a sessionid is specified in a request, fail if we don't have the session.
13913 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13915 * gst/rtsp-server/rtsp-media.c:
13916 Add better debug info
13917 Add some better debug info.
13919 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13921 * examples/test-video.c:
13923 Add support for session timeouts in the example.
13925 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13927 * gst/rtsp-server/rtsp-session-pool.c:
13928 * gst/rtsp-server/rtsp-session-pool.h:
13929 Pass GTimeVal around for performance reasons
13930 Get the current time only once and pass it around so that sessions don't have to
13931 get the current time anymore.
13932 Add experimental support for a GSource that dispatches when the session needs to
13935 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13937 * gst/rtsp-server/rtsp-session.c:
13938 * gst/rtsp-server/rtsp-session.h:
13939 Add better support for session timeouts
13940 Add a method to request the number of milliseconds when a session will timeout.
13942 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13944 * gst/rtsp-server/rtsp-media.c:
13945 * gst/rtsp-server/rtsp-media.h:
13946 Add suport for RTP manager monitoring
13947 Add the first stage in monitoring the rtp manager.
13948 Make sure we don't update the state to something we don't want.
13950 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13952 * gst/rtsp-server/rtsp-client.c:
13953 Add support for session keepalive
13954 Get and update the session timeout for all requests. get the session as early as
13957 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13959 * gst/rtsp-server/rtsp-media-factory.h:
13960 * gst/rtsp-server/rtsp-media.c:
13961 * gst/rtsp-server/rtsp-media.h:
13962 Handle media bus messages
13963 Handle media bus messages in a custom mainloop and dispatch them to the
13964 RTSPMedia objects. Let the default implementation handle some common messages.
13966 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13968 * gst/rtsp-server/rtsp-client.c:
13969 * gst/rtsp-server/rtsp-session-pool.c:
13970 * gst/rtsp-server/rtsp-session.c:
13971 Some more session timeout handling
13972 Move the session header setting code to a central place so that we always add
13973 the timeout parameter too.
13974 Handle timeouts by running the session cleanup code.
13975 Stop media before cleaning up.
13977 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13979 * gst/rtsp-server/rtsp-client.c:
13980 * gst/rtsp-server/rtsp-client.h:
13981 Add timeout property
13982 Add a timeout property ot the client and make the other properties into GObject
13985 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13987 * gst/rtsp-server/rtsp-session-pool.c:
13988 Use getters and setters in property code
13989 Use the getters and setters for the timeout property instead of locking
13992 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13994 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
13996 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13998 * gst/rtsp-server/rtsp-session-pool.c:
13999 * gst/rtsp-server/rtsp-session-pool.h:
14000 * gst/rtsp-server/rtsp-session.c:
14001 * gst/rtsp-server/rtsp-session.h:
14002 Add more timeout stuff
14003 Add method to check if a session is expired.
14004 Add method to perform cleanup on a session pool.
14006 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14008 * gst/rtsp-server/rtsp-client.c:
14009 * gst/rtsp-server/rtsp-session-pool.c:
14010 * gst/rtsp-server/rtsp-session-pool.h:
14011 * gst/rtsp-server/rtsp-session.c:
14012 * gst/rtsp-server/rtsp-session.h:
14013 Add beginnings of session timeouts and limits
14014 Add the timeout value to the Session header for unusual timeout values.
14015 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14016 limit on the amount of retry we do after a sessionid collision.
14017 Add properties to the sessionid and the timeout of a session. Keep track of
14018 creation time and last access time for sessions.
14020 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14022 * gst/rtsp-server/rtsp-client.c:
14023 * gst/rtsp-server/rtsp-media.c:
14024 * gst/rtsp-server/rtsp-media.h:
14025 * gst/rtsp-server/rtsp-sdp.c:
14026 * gst/rtsp-server/rtsp-session-pool.c:
14027 * gst/rtsp-server/rtsp-session.c:
14028 * gst/rtsp-server/rtsp-session.h:
14029 Cleanup of sessions and more
14030 Fix the refcounting of media and sessions in the client. Properly clean up the
14031 session data when the client performs a teardown.
14032 Add Server header to responses.
14033 Allow for multiple uri setups in one session.
14034 Add Range header to the PLAY response and add the range attribute to the SDP
14036 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14037 give the ownership of the sessionid to the session object.
14039 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14041 * gst/rtsp-server/rtsp-server.c:
14042 * gst/rtsp-server/rtsp-server.h:
14044 Rename the 'server_port' variable to simply 'port'.
14046 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14049 * gst/rtsp-server/rtsp-client.c:
14050 * gst/rtsp-server/rtsp-media.c:
14051 * gst/rtsp-server/rtsp-media.h:
14052 * gst/rtsp-server/rtsp-session.c:
14053 * gst/rtsp-server/rtsp-session.h:
14054 Rework the way we handle transports for streams
14055 Make the media accept an array of transports for the streams that we have
14056 configured for the play/pause requests.
14057 Implement server states for a client and its media.
14058 Require 0.10.22.1 (git HEAD) of gstreamer.
14060 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14062 * gst/rtsp-server/rtsp-client.c:
14063 * gst/rtsp-server/rtsp-media-factory.c:
14064 Drop const from functions dealing with urls
14065 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14066 have the right const in them.
14068 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14070 * gst/rtsp-server/rtsp-client.c:
14071 * gst/rtsp-server/rtsp-media.c:
14072 * gst/rtsp-server/rtsp-sdp.c:
14076 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14078 * gst/rtsp-server/rtsp-client.c:
14079 * gst/rtsp-server/rtsp-media-factory.c:
14080 * gst/rtsp-server/rtsp-media.c:
14081 * gst/rtsp-server/rtsp-media.h:
14083 Don't keep a reference to the GstRTSPMedia in the stream.
14084 Free more things when freeing the GstRTSPMedia.
14086 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14089 * gst/rtsp-server/rtsp-media-factory.c:
14090 * gst/rtsp-server/rtsp-media-factory.h:
14091 * gst/rtsp-server/rtsp-media.c:
14092 * gst/rtsp-server/rtsp-media.h:
14093 * gst/rtsp-server/rtsp-server.c:
14094 * gst/rtsp-server/rtsp-server.h:
14095 More docs and small cleanups
14096 Add some more docs and update the README
14097 Cleanup some method names.
14098 Remove an unneeded idx field in the GstRTSPMediaStream
14100 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14103 * examples/Makefile.am:
14104 * examples/test-readme.c:
14105 Add a README and more example code
14106 Add a README file that contains a small introduction on how to use the server
14107 along with the example code explained in the readme.
14109 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14111 * gst/rtsp-server/rtsp-media.c:
14112 * gst/rtsp-server/rtsp-server.c:
14113 Fix some leaks and change default port
14114 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14115 we finished the initial preroll. If we keep them locked, setting the pipeline to
14116 NULL will not stop and clean up the sources correctly.
14117 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14119 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14121 * gst/rtsp-server/rtsp-session.c:
14122 * gst/rtsp-server/rtsp-session.h:
14123 Cleanups to the session object
14124 Remove some unneeded variables in the session state of a stream such as the
14125 owner media and the server transport.
14126 Get the configuration of a media stream in a session based on the media_stream
14127 in the original object instead of our cached index.
14128 Free more data in the finalize method.
14130 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14132 * gst/rtsp-server/rtsp-client.c:
14133 * gst/rtsp-server/rtsp-client.h:
14134 Cleanups and reuse media from DESCRIBE
14135 Handle thread create errors.
14136 Rename some internal methods to better match what they actually do.
14137 Handle misconfiguration of session_pool and media_mapping gracefully.
14138 Cache the DESCRIBE media and uri in the client connection and reuse them when
14139 we receive a SETUP request in the same connection for the same uri.
14140 Cleanup the client connection object.
14142 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14144 * gst/rtsp-server/rtsp-media-factory.c:
14145 * gst/rtsp-server/rtsp-media-factory.h:
14146 * gst/rtsp-server/rtsp-media.c:
14147 * gst/rtsp-server/rtsp-media.h:
14148 Add shared properties to media and factory
14149 Add the shared property to media.
14150 Implement some simple caching in the factory depending on if the media is shared
14153 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14155 * gst/rtsp-server/rtsp-client.c:
14156 Add a little comment
14157 Add some comment about the content-base header.
14159 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14161 * examples/Makefile.am:
14162 * examples/test-mp4.c:
14163 * examples/test-ogg.c:
14164 * examples/test-video.c:
14165 * gst/rtsp-server/Makefile.am:
14166 * gst/rtsp-server/rtsp-client.c:
14167 * gst/rtsp-server/rtsp-client.h:
14168 * gst/rtsp-server/rtsp-media-factory.c:
14169 * gst/rtsp-server/rtsp-media-factory.h:
14170 * gst/rtsp-server/rtsp-media.c:
14171 * gst/rtsp-server/rtsp-media.h:
14172 * gst/rtsp-server/rtsp-sdp.c:
14173 * gst/rtsp-server/rtsp-sdp.h:
14174 * gst/rtsp-server/rtsp-server.c:
14175 * gst/rtsp-server/rtsp-server.h:
14176 * gst/rtsp-server/rtsp-session.c:
14177 * gst/rtsp-server/rtsp-session.h:
14178 Reorganize things, prepare for media sharing
14179 Added various other test server examples
14180 Move the SDP message generation to a separate helper.
14181 Refactor common code for finding the session.
14182 Add content-base for realplayer compatibility
14183 Clean up request uris before processing for better vlc compatibility.
14184 Move prerolling and pipeline construction to the RTSPMedia object.
14185 Use multiudpsink for future pipeline reuse.
14187 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14190 Back to development
14193 === release 0.10.1 ===
14195 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14198 Make 0.10.1 release
14201 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14203 * bindings/vala/Makefile.am:
14205 Add more directories and files to the dist.
14207 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14209 * bindings/python/Makefile.am:
14210 * bindings/python/rtspserver.override:
14211 Fixed compile error of python bindings
14213 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14215 * bindings/vala/gst-rtsp-server-0.10.vapi:
14216 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14217 Marked values as nullable accordingly
14219 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14221 * bindings/vala/gst-rtsp-server-0.10.vapi:
14222 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14223 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14224 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14225 Updated Vala bindings
14227 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14229 * gst/rtsp-server/rtsp-client.c:
14230 * gst/rtsp-server/rtsp-media-mapping.c:
14231 * gst/rtsp-server/rtsp-media-mapping.h:
14232 * gst/rtsp-server/rtsp-media.h:
14233 * gst/rtsp-server/rtsp-session-pool.h:
14234 Cleanups and doc updates
14235 Add some more documentation and do some minor cleanups here and there.
14237 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14239 * gst/rtsp-server/rtsp-client.c:
14240 * gst/rtsp-server/rtsp-media-factory.c:
14241 * gst/rtsp-server/rtsp-media-factory.h:
14242 * gst/rtsp-server/rtsp-media.c:
14243 * gst/rtsp-server/rtsp-media.h:
14244 * gst/rtsp-server/rtsp-session.c:
14245 * gst/rtsp-server/rtsp-session.h:
14247 Rename GstRTSPMediaBin to GstRTSPMedia
14248 Parse the request url into a GstRTSPUri object and pass this object to the
14249 various handlers and methods that require the uri.
14251 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14255 Add some more docs and remove some old code from the example.
14257 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14259 * gst/rtsp-server/rtsp-client.c:
14260 Handle state change failures better
14261 Handle state change failures better when changing the state of the pipeline to
14264 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14266 * gst/rtsp-server/rtsp-media-factory.c:
14267 * gst/rtsp-server/rtsp-media-factory.h:
14268 Make element creation more extendible
14269 Add get_element vmethod to the default MediaFactory so that subclasses can just
14270 override that method and still use the default logic for making a MediaBin from
14273 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14276 * gst/rtsp-server/Makefile.am:
14277 * gst/rtsp-server/rtsp-client.c:
14278 * gst/rtsp-server/rtsp-client.h:
14279 * gst/rtsp-server/rtsp-media-factory.c:
14280 * gst/rtsp-server/rtsp-media-factory.h:
14281 * gst/rtsp-server/rtsp-media-mapping.c:
14282 * gst/rtsp-server/rtsp-media-mapping.h:
14283 * gst/rtsp-server/rtsp-media.c:
14284 * gst/rtsp-server/rtsp-media.h:
14285 * gst/rtsp-server/rtsp-server.c:
14286 * gst/rtsp-server/rtsp-server.h:
14287 * gst/rtsp-server/rtsp-session.c:
14288 * gst/rtsp-server/rtsp-session.h:
14289 Make the server handle arbitrary pipelines
14290 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14291 The GstMediaBin object has a handle to a bin with elements and to a list of
14292 GstMediaStream objects that this bin produces.
14293 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14294 with methods to register and remove those mappings.
14295 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14296 used by the server instance.
14297 Modify the example application so that it shows how to create custom pipelines
14298 attached to a specific mount point.
14299 Various misc cleanps.
14301 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14303 * gst/rtsp-server/rtsp-server.c:
14304 * gst/rtsp-server/rtsp-server.h:
14305 Allow setting a custom media factory for a server
14307 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14309 * gst/rtsp-server/rtsp-client.c:
14310 * gst/rtsp-server/rtsp-client.h:
14311 Allow setting a custom media factory for a client.
14313 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14315 * gst/rtsp-server/Makefile.am:
14316 Add Makefile entry for the media factory
14318 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14320 * gst/rtsp-server/rtsp-media-factory.c:
14321 * gst/rtsp-server/rtsp-media-factory.h:
14322 Add media factory to map urls to media pipeline objects.
14324 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14326 * gst/rtsp-server/rtsp-media.c:
14327 * gst/rtsp-server/rtsp-media.h:
14328 Add comments. Remove unused field
14330 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14332 * gst/rtsp-server/rtsp-session-pool.c:
14333 * gst/rtsp-server/rtsp-session-pool.h:
14334 Allow custom session pools to override the session id allocation algorithms Add some comments.
14336 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14338 * gst/rtsp-server/rtsp-session.h:
14341 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14343 * gst/rtsp-server/rtsp-client.c:
14344 * gst/rtsp-server/rtsp-client.h:
14345 Move the connection code in one place Add some comments
14347 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14349 * gst/rtsp-server/rtsp-server.c:
14350 * gst/rtsp-server/rtsp-server.h:
14351 Make vmethod to create and accept new clients. Add some docs.
14353 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14355 * gst/rtsp-server/rtsp-server.c:
14356 * gst/rtsp-server/rtsp-server.h:
14357 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14359 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14361 * gst/rtsp-server/rtsp-client.c:
14362 * gst/rtsp-server/rtsp-client.h:
14363 Name the parameters more appropriately.
14365 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14367 * gst/rtsp-server/rtsp-session-pool.c:
14368 Do some more cleanup of the session pool.
14370 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14372 * gst/rtsp-server/Makefile.am:
14373 * gst/rtsp-server/rtsp-client.c:
14374 Check if return value of gst_rtsp_session_get_media is not NULL
14376 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14378 * gst/rtsp-server/Makefile.am:
14379 Install rtsp-session and rtsp-session-pool headers
14381 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14386 * bindings/python/Makefile.am:
14387 * bindings/python/arg-types.py:
14388 * bindings/python/codegen/Makefile.am:
14389 * bindings/python/codegen/__init__.py:
14390 * bindings/python/codegen/argtypes.py:
14391 * bindings/python/codegen/code-coverage.py:
14392 * bindings/python/codegen/codegen.py:
14393 * bindings/python/codegen/definitions.py:
14394 * bindings/python/codegen/defsparser.py:
14395 * bindings/python/codegen/docextract.py:
14396 * bindings/python/codegen/docgen.py:
14397 * bindings/python/codegen/fileprefix.override:
14398 * bindings/python/codegen/fileprefixmodule.c:
14399 * bindings/python/codegen/h2def.py:
14400 * bindings/python/codegen/mergedefs.py:
14401 * bindings/python/codegen/mkskel.py:
14402 * bindings/python/codegen/override.py:
14403 * bindings/python/codegen/reversewrapper.py:
14404 * bindings/python/codegen/scmexpr.py:
14405 * bindings/python/rtspserver-types.defs:
14406 * bindings/python/rtspserver.defs:
14407 * bindings/python/rtspserver.override:
14408 * bindings/python/rtspservermodule.c:
14410 Add python bindings.
14412 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14414 * bindings/Makefile.am:
14416 Don't go into python dir when requirements for python bindings are missing
14418 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14420 * bindings/Makefile.am:
14421 * bindings/vala/Makefile.am:
14423 Install Vala bindings if vala is available
14425 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14427 * bindings/vala/gst-rtsp-server-0.10.deps:
14428 * bindings/vala/gst-rtsp-server-0.10.vapi:
14429 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14430 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14431 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14432 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14433 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14434 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14435 Regenerated Vala bindings
14437 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14439 * bindings/vala/gst-rtsp-server.vapi:
14440 * bindings/vala/packages/gst-rtsp-server.metadata:
14441 Fixed typo in included headers for vala bindings
14443 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14447 * pkgconfig/Makefile.am:
14448 * pkgconfig/gst-rtsp-server.pc.in:
14449 Added pkgconfig file
14451 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14453 * bindings/vala/gst-rtsp-server.vapi:
14454 * bindings/vala/packages/gst-rtsp-server.excludes:
14455 * bindings/vala/packages/gst-rtsp-server.gi:
14456 * bindings/vala/packages/gst-rtsp-server.metadata:
14457 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14459 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14461 * bindings/vala/gst-rtsp-server.vapi:
14462 * bindings/vala/packages/gst-rtsp-server.deps:
14463 * bindings/vala/packages/gst-rtsp-server.files:
14464 * bindings/vala/packages/gst-rtsp-server.gi:
14465 * bindings/vala/packages/gst-rtsp-server.metadata:
14466 * bindings/vala/packages/gst-rtsp-server.namespace:
14467 Added Vala bindings
14469 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14471 * gst/rtsp-server/rtsp-session.c:
14472 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14474 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14476 * examples/Makefile.am:
14477 * gst/rtsp-server/Makefile.am:
14478 Put GStreamer version in library name
14480 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14482 * examples/Makefile.am:
14483 * gst/rtsp-server/Makefile.am:
14484 Fix some issues to pass distcheck
14486 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14488 * gst/rtsp-server/rtsp-server.c:
14489 Added port property to GstRTSPServer class.
14491 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14496 * examples/Makefile.am:
14499 * gst/rtsp-server/Makefile.am:
14500 * gst/rtsp-server/rtsp-client.c:
14501 * gst/rtsp-server/rtsp-client.h:
14502 * gst/rtsp-server/rtsp-media.c:
14503 * gst/rtsp-server/rtsp-media.h:
14504 * gst/rtsp-server/rtsp-server.c:
14505 * gst/rtsp-server/rtsp-server.h:
14506 * gst/rtsp-server/rtsp-session-pool.c:
14507 * gst/rtsp-server/rtsp-session-pool.h:
14508 * gst/rtsp-server/rtsp-session.c:
14509 * gst/rtsp-server/rtsp-session.h:
14511 Split in library and example program
14513 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14515 * src/rtsp-client.h:
14516 Removed obsolete variable
14518 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14520 * src/rtsp-client.c:
14521 * src/rtsp-client.h:
14522 Removed pipeline variable GstRTSPClient, because it's only used in one function
14524 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14526 * src/rtsp-media.c:
14527 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14529 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14531 * src/rtsp-session.c:
14532 Initialize some more vars.
14534 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14536 * src/rtsp-session.c:
14537 Initialize variable to avoid compiler warning.
14539 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14542 Add a reasonable generic .gitignore