3 2022-02-03 19:53:25 +0000 Tim-Philipp Müller <tim@centricular.com>
7 * docs/gst_plugins_cache.json:
8 * gst-rtsp-server.doap:
12 2022-02-03 19:53:18 +0000 Tim-Philipp Müller <tim@centricular.com>
15 Update ChangeLogs for 1.20.0
17 === release 1.19.90 ===
19 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
24 * docs/gst_plugins_cache.json:
25 * gst-rtsp-server.doap:
29 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
32 Update ChangeLogs for 1.19.90
34 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
36 * examples/test-appsrc2.c:
37 gst-rtsp-server: Fix leak in appsrc2 example
38 In the need-data appsrc callback, a buffer is pulled from the
39 appsink. This buffer is then copied so that metadata is writable.
40 The copy is pushed to the appsrc but it doesn't take ownership
41 of the buffer so we need to manually unref it. The original buffer
42 is finally unreffed when the sample is freed.
43 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
45 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
49 meson: Add explicit check: kwarg to all run_command() calls
50 This is required since Meson 0.61.0, and causes a warning to be
52 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
53 https://github.com/mesonbuild/meson/issues/9300
54 This exposed a bunch of places where we had broken run_command()
55 calls, unnecessary run_command() calls, and places where check: true
57 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
59 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
61 * gst/rtsp-server/meson.build:
62 rtsp-server: add gst_dep to gst_rtsp_server_deps
63 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
64 will avoid the following build failure, because the correct girdir
65 location will be retrieved from gstreamer-1.0.pc:
66 /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
67 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
68 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
69 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
70 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
72 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
73 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
75 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
77 * gst/rtsp-server/rtsp-stream.c:
78 rtsp-stream: fix get_rates raciness
79 Prior to this patch, we considered that a stream was blocking
80 whenever a pad probe was triggered for either the RTP pad or
82 This led to situations where we subsequently unblocked and expected
83 to find a segment on the RTP pad, which was racy.
84 Instead, we now only consider that the stream is blocking when
85 the pad probe for the RTP pad has triggered with a blockable object
86 (buffer, buffer list, gap event).
87 The RTCP pad is simply blocked without affecting the state of the
90 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
92 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
94 * docs/gst_plugins_cache.json:
98 === release 1.19.3 ===
100 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
105 * docs/gst_plugins_cache.json:
106 * gst-rtsp-server.doap:
110 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
113 Update ChangeLogs for 1.19.3
115 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
118 meson: require matching GStreamer dep versions for unstable development releases
119 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
120 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
122 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
124 * tests/check/meson.build:
125 meson: update for meson.build_root() and .build_source() deprecation
126 -> use meson.project_build_root() or .global_build_root() instead.
127 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
129 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
132 * tests/check/meson.build:
133 meson: update for dep.get_pkgconfig_variable() deprecation
134 ... in favour of dep.get_variable('foo', ..) which in some
135 cases allows for further cleanups in future since we can
136 extract variables from pkg-config dependencies as well as
137 internal dependencies using this mechanism.
138 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
140 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
142 * gst/rtsp-server/meson.build:
143 * gst/rtsp-sink/meson.build:
144 rtsp-server: define G_LOG_DOMAIN
146 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
148 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
151 meson: bump meson requirement to >= 0.59
152 For monorepo build and ugly/bad, for advanced feature
153 option API like get_option('xyz').required(..) which
154 we use in combination with the 'gpl' option.
155 For rest of modules for consistency (people will likely
156 use newer features based on the top-level requirement).
157 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
159 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
162 meson: Streamline the way we detect when to build documentation
163 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
165 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
168 * gst/rtsp-server/meson.build:
170 meson: List libraries and their corresponding gir definition
171 Introduces a `libraries` variable that contains all libraries in a
172 list with the following format:
176 'lib': library_object
177 'gir': [ {full gir definition in a dict } ]
182 It therefore refactors the way we build the gir so that we can reuse the
183 same information to build them against 'gstreamer-full' in gst-build
184 when linking statically
185 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
187 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
189 * gst/rtsp-server/meson.build:
190 meson: Mark files as files()
191 Making it more robust and future proof
192 And fix issues that it creates
193 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
195 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
197 * gst/rtsp-server/rtsp-media.c:
198 rtsp-media: Unprepare suspended medias too
199 Previously suspended medias immediately reached the UNPREPARED state
200 without going through the media's unprepare() vfunc. This didn't allow
201 the media subclass to do any additional cleanup, and for example the
202 shutdown-eos property of GstRTSPMedia was ignored.
203 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
205 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
207 * gst/rtsp-server/rtsp-media.c:
208 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
209 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
211 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
213 * gst/rtsp-server/rtsp-client.c:
214 * gst/rtsp-server/rtsp-session.c:
215 * gst/rtsp-server/rtsp-session.h:
216 rtsp-client: make sure sessmedia will not get freed while used
217 handle_*_request() functions were all retrieving the session media from
218 the session by calling gst_rtsp_session_get_media () which is a transfer-none
219 call. If a session timeout happens at that time, the session media may get freed
220 making the pointer invalid..
222 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
224 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
226 * gst/rtsp-server/rtsp-media.c:
227 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
228 Previously the status was only changed for other medias.
229 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
231 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
233 * gst/rtsp-server/rtsp-session.c:
234 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
235 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
236 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
238 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
241 doc: update IRC links to OFTC
242 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
244 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
246 * docs/gst_plugins_cache.json:
249 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
251 === release 1.19.2 ===
253 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
258 * docs/gst_plugins_cache.json:
259 * gst-rtsp-server.doap:
263 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
265 * gst/rtsp-server/rtsp-media.c:
266 * gst/rtsp-server/rtsp-stream.c:
267 * gst/rtsp-server/rtsp-stream.h:
268 * gst/rtsp-sink/gstrtspclientsink.c:
269 Protection against early RTCP packets.
270 When receiving RTCP packets early the funnel is not ready yet and
271 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
272 This causes the thread that handle RTCP packets to go to pause mode.
273 Since this thread is in pause mode there will be no further callbacks to
274 handle keep-alive for incoming RTCP packets. This will make the session
275 time out if the client is not using another keep-alive mechanism.
276 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
277 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
279 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
283 Update COPYING.LIB, COPYING files
284 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
286 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
288 * docs/gst_plugins_cache.json:
292 === release 1.19.1 ===
294 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
299 * docs/gst_plugins_cache.json:
300 * gst-rtsp-server.doap:
304 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
306 * gst/rtsp-server/rtsp-stream.c:
307 rtsp-stream: use new gst_buffer_new_memdup()
308 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
310 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
312 * gst/rtsp-server/rtsp-media-factory-uri.c:
313 rtsp-media: fix leak when adding converter
314 Free the previous caps before reusing the variable for the converter caps.
315 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
317 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
319 * gst/rtsp-server/rtsp-client.c:
320 rtsp-client: fix leak adding headers
321 gst_rtsp_message_add_header() makes a copy of the header, instead
323 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
325 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
327 * gst/rtsp-server/rtsp-stream.c:
328 Use gst_element_request_pad_simple...
329 Instead of the deprecated gst_element_get_request_pad.
330 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
332 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
334 * gst/rtsp-server/rtsp-media.c:
335 rtsp-media: Ensure the bus watch is removed during unprepare
336 It's possible for the destruction of the source to be delayed.
337 Instead of relying on the dispose() to remove the bus watch, do
339 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
341 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
344 docs: minor spelling correction in README
345 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
347 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
349 * examples/test-replay-server.c:
350 test-replay-server: minor spelling corrections
351 Bumped on these while investigating the example code.
352 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
354 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
356 * tests/check/gst/stream.c:
357 tests: Don't fail tests if IPv6 not available.
358 On computers with IPv6 disabled it shouldn't result in a test failure.
359 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
361 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
363 * gst/rtsp-server/rtsp-media.c:
364 rtsp-media: Add one more case to seek avoidance
365 This is an extension to the previous commit. There can also be cases where the
366 start position is not specified, in those cases we should also avoid doing
367 seeking unless it's forced.
368 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
370 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
372 * gst/rtsp-server/rtsp-media.c:
373 rtsp-media: Improve skipping trickmode seek.
374 We can also skip the seek if the end range is already
376 Avoids initial seek on play start if playing full stream.
377 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
379 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
381 * gst/rtsp-sink/gstrtspclientsink.c:
382 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
383 It's sufficient to run them during the FIRST stage instead of in both.
384 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
386 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
388 * tests/check/gst/rtspclientsink.c:
389 tests: rtspclientsink: fix some leaks
390 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
392 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
394 * gst/rtsp-sink/gstrtspclientsink.c:
395 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
396 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
398 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
400 * tests/check/gst/rtspclientsink.c:
401 rtspclientsink: add unit test for potential shutdown deadlock
402 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
404 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
406 * gst/rtsp-sink/gstrtspclientsink.c:
407 rtspclientsink: fix deadlock on shutdown before preroll
408 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
409 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
411 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
413 * gst/rtsp-server/rtsp-stream.c:
414 rtsp-stream: avoid deadlock in send_func
415 Currently the send_func() runs in a thread of its own which is started
416 the first time we enter handle_new_sample(). It runs in an outer loop
417 until priv->continue_sending is FALSE, which happens when a TEARDOWN
418 request is received. We use a local variable, cont, which is initialized
419 to TRUE, meaning that we will always enter the outer loop, and at the
420 end of the outer loop we assign it the value of priv->continue_sending.
421 Within the outer loop there is an inner loop, where we wait to be
422 signaled when there is more data to send. The inner loop is exited when
423 priv->send_cookie has changed value, which it does when more data is
424 available or when a TEARDOWN has been received.
425 But if we get a TEARDOWN before send_func() is entered we will get stuck
426 in the inner loop because no one will increase priv->session_cookie
428 By not entering the outer loop in send_func() if priv->continue_sending
429 is FALSE we make sure that we do not get stuck in send_func()'s inner
430 loop should we receive a TEARDOWN before the send thread has started.
431 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
432 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
434 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
436 * gst/rtsp-server/rtsp-client.c:
437 rtsp-client: cleanup transports during TEARDOWN
438 When tunneling RTP over RTSP the stream transports are stored in a hash
439 table in the GstRTSPClientPrivate struct. They are used for, among other
440 things, mapping channel id to stream transports when receiving data from
441 the client. The stream tranports are created and added to the hash table
442 in handle_setup_request(), but unfortuately they are not removed in
443 handle_teardown_request(). This means that if the client sends data on
444 the RTSP connection after it has sent the TEARDOWN, which is often the
445 case when audio backchannel is enabled, handle_data() will still be able
446 to map the channel to a session transport and pass the data along to it.
447 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
448 because the stream is no longer joined to a bin.
449 We avoid this by removing the stream transports from the hash table when
450 we handle the TEARDOWN request.
451 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
453 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
455 * docs/gst_plugins_cache.json:
456 * gst/rtsp-sink/gstrtspclientsink.c:
457 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
458 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
460 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
462 * tests/check/gst/client.c:
463 Add test cases for mountpoint of '/'
464 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
466 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
468 * gst/rtsp-server/rtsp-client.c:
469 * gst/rtsp-server/rtsp-mount-points.c:
470 * gst/rtsp-server/rtsp-session-media.c:
471 Make a mount point of "/" work correctly.
472 As far as I can tell, this is neither explicitly allowed nor
473 forbidden by RFC 7826.
474 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
475 use in the wild (presumably with non-GStreamer servers).
476 GStreamer's prior behavior was confusing, in that
477 gst_rtsp_mount_points_add_factory() would appear to accept a mount
478 path of "" or "/", but later connection attempts would fail with a
479 "media not found" error.
480 This commit makes a mount path of "/" work for either form of URL,
481 while an empty mount path ("") is rejected and logs a warning.
482 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
484 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
486 * docs/gst_plugins_cache.json:
487 * gst/rtsp-sink/gstrtspclientsink.c:
488 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
489 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
491 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
493 * gst/rtsp-server/rtsp-media.c:
494 rtsp-media: Only count senders when counting blocked streams
495 Only sender streams sends the GstRTSPStreamBlocking message, so only
496 these should be counted before setting media status to prepared.
497 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
499 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
501 * gst/rtsp-sink/gstrtspclientsink.c:
502 rtspclientsink add proper support for uri queries
503 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
505 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
507 * gst/rtsp-server/rtsp-client.c:
508 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
509 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
510 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
512 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
514 * gst/rtsp-server/rtsp-stream.c:
515 rtsp-stream: collect a clock_rate when blocking
516 This lets us provide a clock_rate in a fashion similar to the
517 other code paths in get_rtpinfo()
518 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
520 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
522 * gst/rtsp-server/rtsp-media.c:
523 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
524 Otherwise this will cause memory corruption as the property expects a 64
526 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
528 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
530 * gst/rtsp-server/rtsp-media.c:
531 * gst/rtsp-server/rtsp-stream.c:
532 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
533 To prevent cases with prerolling when the inactive stream prerolls first
534 and the server proceeds without waiting for the active stream, we will
535 ignore GstRTSPStreamBlocking messages from incomplete streams. When
536 there are no complete streams (during DESCRIBE), we will listen to all
538 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
540 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
542 * tests/check/gst/media.c:
543 * tests/check/meson.build:
544 * tests/files/test.avi:
545 media test: Add test for seeking one active stream with a demuxer
546 Add another seek_one_active_stream test but with a demuxer. The demuxer
547 will flush both streams in opposed to the existing test which only
548 flushes the active stream. This will help exposing problems with the
549 prerolling process after a flushing seek.
550 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
552 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
554 * gst/rtsp-server/meson.build:
556 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
557 * pkgconfig/gstreamer-rtsp-server.pc.in:
558 * pkgconfig/meson.build:
559 Meson: Use pkg-config generator
560 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
562 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
565 meson: update glib minimum version to 2.56
566 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
568 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
570 * examples/test-launch.c:
571 * gst/rtsp-server/rtsp-media-factory.c:
572 * gst/rtsp-server/rtsp-media-factory.h:
573 * gst/rtsp-server/rtsp-media.c:
574 * gst/rtsp-server/rtsp-server-internal.h:
575 * gst/rtsp-server/rtsp-stream.c:
576 * tests/check/gst/client.c:
577 rtsp-media-factory: expose API to disable RTCP
578 This is supported by the RFC, and can be useful on systems where
579 allocating two consecutive ports is problematic, and RTCP is not
581 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
583 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
585 * hooks/pre-commit.hook:
587 git: use our standard pre commit hook
588 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
590 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
592 * gst/rtsp-server/rtsp-stream.c:
593 rtsp-stream: make use of blocked_running_time in query_position
594 When blocking, the sink element will not have received a buffer
595 yet and the position query will fail. Instead, we make use of
596 the running time of the buffer we blocked on.
597 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
599 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
601 * gst/rtsp-server/rtsp-stream.c:
602 rtsp-stream: collect rtp info when blocking
603 We don't unblock the stream anymore before replying to the
604 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
605 so the sinks don't have a last-sample after potentially flush
606 seeking. seek_trickmode waits for preroll however, which means
607 the stream will block and wait for a first buffer. Subsequent
608 calls to get_rtpinfo() can thus make use of the information.
609 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
610 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
612 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
614 * examples/meson.build:
615 * examples/test-replay-server.c:
616 * examples/test-replay-server.h:
617 examples: Add an example for loop playback
618 This demo example shows a way of file loop playback of a given source.
619 Note that client seek request is not properly implemented yet.
620 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
622 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
624 * gst/rtsp-server/rtsp-media.c:
625 rtsp-media: Plug memory leak
626 The get-storage signal of rtpbin increases the ref count of the storage.
627 So we have to unref it after usage.
628 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
630 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
632 * gst/rtsp-server/rtsp-media.c:
633 rtsp-media: Get rates only on sender streams
634 When play a media with both sender and receiver stream, like ONVIF
635 back channel audio in, gst_rtsp_media_get_rates call
636 gst_rtsp_stream_get_rates for each stream to set the rates. But
637 gst_rtsp_stream_get_rates return false for the receiver steam, which
638 lead a g_assert crash.
639 Instead to get rates on all streams, now just get rates on sender
641 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
643 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
645 * gst/rtsp-server/rtsp-media.c:
646 * gst/rtsp-server/rtsp-server-internal.h:
647 * gst/rtsp-server/rtsp-stream.c:
648 rtsp-media: set a 0 storage size for TCP receivers
649 ulpfec correction is obviously useless when receiving a stream
650 over TCP, and in TCP modes the rtp storage receives non
651 timestamped buffers, causing it to queue buffers indefinitely,
652 until the queue grows so large that sanity checks kick in and
653 warnings start to get emitted.
654 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
656 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
658 * gst/rtsp-server/rtsp-stream.c:
659 rtsp-stream: preroll on gap events
660 This allows negotiating a SDP with all streams present, but only
661 start sending packets at some later point in time
662 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
664 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
666 * gst/rtsp-server/rtsp-media.c:
667 rtsp-media: do not unblock on unsuspend
668 rtsp_media_unsuspend() is called from handle_play_request()
669 before sending the play response. Unblocking the streams here
670 was causing data to be sent out before the client was ready
671 to handle it, with obvious side effects such as initial packets
672 getting discarded, causing decoding errors.
673 Instead we can simply let the media streams be unblocked when
674 the state of the media is set to PLAYING, which occurs after
675 sending the play response.
676 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
678 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
681 ci: include template from gst-ci master branch again
683 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
685 * docs/gst_plugins_cache.json:
689 === release 1.18.0 ===
691 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
697 * docs/gst_plugins_cache.json:
698 * gst-rtsp-server.doap:
702 === release 1.17.90 ===
704 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
709 * docs/gst_plugins_cache.json:
710 * gst-rtsp-server.doap:
714 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
716 * gst/rtsp-server/rtsp-thread-pool.c:
717 rtsp-thread-pool.c: fix clang 10 warning
718 clang 10 is complaining about incompatible types due to the
721 ../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]
723 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
725 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
727 * gst/rtsp-server/rtsp-thread-pool.c:
728 rtsp-thread-pool.c: fix clang 10 warning
729 clang 10 is complaining about incompatible types due to the
732 ../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]
734 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
736 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
738 * gst/rtsp-server/rtsp-sdp.c:
739 rtsp-sdp: Fix resource leak in mikey messsage
740 Fixed a resource leak for mikey message while adding crypto session
742 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
744 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
747 * scripts/extract-release-date-from-doap-file.py:
748 meson: set release date from .doap file for releases
749 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
751 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
753 * gst/rtsp-server/rtsp-stream.c:
754 rtsp-stream: explicitly set caps on udpsrc elements
755 This causes them to send caps events before data flow, which is
756 usually a pretty correct thing to do!
757 Not doing so manifested in a bug where ssrcdemux wouldn't forward
758 the caps it had received with an extra ssrc field, as it hadn't
759 received any caps event.
761 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
763 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
765 * docs/gst_plugins_cache.json:
769 === release 1.17.2 ===
771 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
776 * docs/gst_plugins_cache.json:
777 * gst-rtsp-server.doap:
781 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
783 * docs/gst_plugins_cache.json:
784 doc: Stop documenting properties from parents
786 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
788 * docs/gst_plugins_cache.json:
789 docs: Fix version in the plugins cache
790 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
792 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
794 * gst/rtsp-sink/gstrtspclientsink.c:
795 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
796 It's deprecated, unneeded and doesn't do anything anymore.
797 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
799 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
804 === release 1.17.1 ===
806 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
811 * docs/gst_plugins_cache.json:
812 * gst-rtsp-server.doap:
816 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
818 * gst/rtsp-server/rtsp-media.c:
819 rtsp-media: Add/configure transports when completing the pipeline
820 Otherwise the transports are not set up yet during the PLAY request
821 handling when unsuspending (and thus unblocking) the media.
822 In case of live pipelines this then causes the first few packets to go
823 to the sinks before they know what to do with them, and they simply
824 discard them which is rather suboptimal in case of keyframes.
825 For non-live pipelines this is not a problem because the sink will still
826 be PAUSED and as such not send out the data yet but wait until it goes
827 to PLAYING, which is late enough.
828 Adding the transports multiple times is not a problem: if the transport
829 is already added it won't be added another time and TRUE will be
831 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
833 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
834 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
836 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
838 * gst/rtsp-server/rtsp-media.c:
839 rtsp-media: Fix misleading comment
840 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
842 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
844 * gst/rtsp-server/rtsp-media.c:
845 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
846 The pad probes are not needed anymore at this point and later when
847 reaching buffering 100% only the state is changed, no unblocking
849 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
851 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
853 * gst/rtsp-server/rtsp-media.c:
854 rtsp-media: Remove duplicated media_unblock() function
855 It does literally the same as media_streams_set_blocked(FALSE).
856 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
858 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
860 * examples/test-onvif-server.c:
861 test-onvif-server: cast ntp-offset property value to 64 bit
862 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
864 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
866 * docs/gst_plugins_cache.json:
867 docs: Update plugins cache
869 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
871 * examples/test-onvif-server.c:
872 * examples/test-onvif-server.h:
873 * gst/rtsp-server/rtsp-onvif-media-factory.h:
874 onvif-media-factory: define autoptr cleanup function
875 And have the factory in the onvif-server example inherit from
876 GstRTSPOnvifMediaFactory.
877 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
879 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
881 * docs/gst_plugins_cache.json:
882 docs: Update plugins cache
884 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
886 * tests/check/gst/rtspserver.c:
887 tests: enforce I420 format
888 Test was not enforcing a video format on videotestsrc. I420 was picked as it
889 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
890 true (gst-plugins-base!689).
891 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
893 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
895 * gst/rtsp-sink/gstrtspclientsink.c:
896 plugins: uddate gst_type_mark_as_plugin_api() calls
898 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
901 doc: Require hotdoc >= 0.11.0
903 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
905 * docs/gst_plugins_cache.json:
906 docs: Update gst_plugins_cache.json
908 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
910 * gst/rtsp-sink/gstrtspclientsink.c:
911 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
913 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
915 * gst/rtsp-server/meson.build:
916 meson: gir: remove bogus sources_top_dir kwarg
917 Doesn't actually exist. Was fixed differently in Meson
918 so that the user doesn't have to specify it.
919 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
921 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
923 * tests/check/meson.build:
924 tests: put registry into tests/check not the gst/ subdir
925 Underscorify the test name before setting GST_REGISTRY,
926 so the registry actually ends up in the current build dir
928 For consistency with the other modules, but should also
929 avoid problems on windows.
930 Also fix indentation of environment block.
931 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
933 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
935 * tests/check/meson.build:
936 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
937 If core is built as a subproject (e.g. as in gst-build), make sure to use
938 the gst-plugin-scanner from the built subproject. Without this, gstreamer
939 might accidentally use the gst-plugin-scanner from the install prefix if
940 that exists, which in turn might drag in gst library versions we didn't
941 mean to drag in. Those gst library versions might then be older than
942 what our current build needs, and might cause our newly-built plugins
943 to get blacklisted in the test registry because they rely on a symbol
944 that the wrongly-pulled in gst lib doesn't have.
945 This should fix running of unit tests in gst-build when invoking
946 meson test or ninja test from outside the devenv for the case where
947 there is an older or different-version gst-plugin-scanner installed
948 in the install prefix.
949 In case no gst-plugin-scanner is installed in the install prefix, this
950 will fix "GStreamer-WARNING: External plugin loader failed. This most
951 likely means that the plugin loader helper binary was not found or
952 could not be run. You might need to set the GST_PLUGIN_SCANNER
953 environment variable if your setup is unusual." warnings when running
955 In the case where we find GStreamer core via pkg-config we use
956 a newly-added pkg-config var "pluginscannerdir" to get the right
957 directory. This has the benefit of working transparently for both
958 installed and uninstalled pkg-config files/setups.
959 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
961 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
963 * tests/check/meson.build:
964 tests: gst-plugins-base and -bad plugins are required for the unit tests
965 Make hard requirement until we have more fine-grained control
966 in the unit tests. Of course the presence of the .pc file doesn't
967 imply that the plugins we need are actually there, but it's at
968 least a step in the right direction.
969 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
971 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
973 * tests/check/meson.build:
974 tests: pick up rtsp-server plugins from build directory only
975 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
977 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
979 * gst/rtsp-server/rtsp-media.c:
980 rtsp-media: wait for all GstRTSPStreamBlocking messages
981 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
982 each active stream when checking if all streams are blocked.
983 Without this change there will be a race condition when using two or
984 more streams and rtsp-media receives a GstRTSPStreamBlocking message
985 from one of the streams. This is because rtsp-media then checks if all
986 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
987 stream. This function call returns TRUE if the stream has sent a
988 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
989 receive this message. This would then result in that rtsp-media
990 erroneously thinks it is blocking all streams which could result in
991 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
992 preroll, this could result in that rtsp-media thinks that the pipeline
993 is prerolled even though that might not be the case.
994 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
996 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
998 * gst/rtsp-server/rtsp-media.c:
999 rtsp-media: update expected_async_done during suspend
1000 Set expected_async_done to FALSE in default_suspend() if a state change
1001 occurs and the return value from set_target_state() is something other
1002 than GST_STATE_CHANGE_ASYNC.
1003 Without this change there is a risk that expected_async_done will be
1004 TRUE even though no asynchronous state change is taking place. This
1005 could happen if the pipeline is set to PAUSED using
1006 media_set_pipeline_state_locked(), an asynchronous state change starts
1007 and then the media is suspended (which could result in a state change,
1008 aborting the asynchronous state change). If the media is suspended
1009 before the asynchronous state change ends then expected_async_done will
1010 be TRUE but no asynchronous state change is taking place.
1011 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
1013 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
1015 * gst/rtsp-server/rtsp-client.c:
1016 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1017 There was a race condition where client was being finalized and
1018 concurrently in some other thread the rtsp ctrl timout was relying on
1019 client data that was being freed.
1020 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1021 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1023 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1025 * gst/rtsp-server/rtsp-media-factory.c:
1026 * gst/rtsp-server/rtsp-media-factory.h:
1027 * gst/rtsp-server/rtsp-media.c:
1028 * gst/rtsp-server/rtsp-media.h:
1029 media-factory: complete DSCP QoS setting support
1030 add dscp_qos setting support at factory and media level to setup IP DSCP
1031 field of bounded UDP sinks.
1032 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1033 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1035 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1037 * gst/rtsp-server/rtsp-client.c:
1038 rtsp-client: Fix some race conditions around timeout source removal
1039 We always need to take the lock while accessing it as otherwise another
1040 thread might've removed it in the meantime. Also when destroying and
1041 creating a new one, ensure that the mutex is not shortly unlocked in
1042 between as during that time another one might potentially be created
1044 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1046 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1048 * gst/rtsp-server/rtsp-media.c:
1049 * gst/rtsp-server/rtsp-stream.c:
1050 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1051 And the same for gst_rtsp_stream_get_rates().
1052 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1054 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1056 * examples/test-onvif-server.c:
1057 examples: test-onvif-server: fix compiler warnings on raspbian
1058 Fix printf format for 64-bit variables.
1059 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1061 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1063 * gst/rtsp-server/rtsp-stream-transport.c:
1064 * gst/rtsp-server/rtsp-stream-transport.h:
1065 * gst/rtsp-server/rtsp-stream.c:
1066 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1067 The old API is preserved now and new API was added that provides the
1068 additional parameter to the callback.
1069 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1070 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1072 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1074 * gst/rtsp-server/rtsp-client.c:
1075 rtsp-client: Store the timeout source by pointer instead of id
1076 That way we don't have to retrieve it again from the main context when
1077 destroying it but can directly do so.
1078 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1080 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1082 * gst/rtsp-server/rtsp-client.c:
1083 rtsp-client: Clean up watch/watch context and related state consistently
1084 And assert that it was cleaned up properly before the client is
1085 finalized. If something is still around when the client is shut down
1086 then something went very wrong before.
1087 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1089 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1091 * gst/rtsp-server/rtsp-client.c:
1092 * tests/check/gst/rtspserver.c:
1093 rtsp-client: Combine the pre-session and post-session timeout
1094 They previously used the same state but different mechanisms and
1095 functions, which was difficult to follow, error prone and simply
1097 Also adjust the test for the post-session timeout a bit to be less racy
1098 now that the timing has slightly changed.
1099 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1101 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1103 * gst/rtsp-server/rtsp-client.c:
1104 rtsp-client: Don't ever close the client connection directly when a session is torn down
1105 There might be other sessions that are running over the same RTSP
1106 connection and we should not simply close the client directly if one of
1108 By default the connection will be closed once the client closes it or
1109 the OS does. This behaviour can be adjusted with the
1110 post-session-timeout property, which allows to close it automatically
1111 from the server side after all sessions are gone and the given timeout
1113 This reverts the previous commit.
1114 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1116 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1118 * gst/rtsp-server/rtsp-client.c:
1119 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1120 Instead of closing it never at all. Previously there was only code that
1121 closed the client asynchronously if sending the response happened
1122 asynchrously at a later time.
1123 Thanks to Christian M for debugging this issue.
1124 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1125 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1127 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1129 * gst/rtsp-server/rtsp-stream.c:
1130 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1131 Otherwise no sink is found for multicast sreams and the less accurate
1132 fallback is used to determine the current sequence number and timestamp.
1134 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1136 * gst/rtsp-server/rtsp-auth.c:
1137 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1138 When using the basic authentication scheme, we wouldn't validate that
1139 the authorization field of the credentials is not NULL and pass it on
1140 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1141 dereference the NULL pointer and crash.
1142 A specially crafted (read: invalid) RTSP header can cause this to
1144 As a solution, check for the authorization to be not NULL before
1145 continuing processing it and if it is simply fail authentication.
1146 This fixes CVE-2020-6095 and TALOS-2020-1018.
1147 Discovered by Peter Wang of Cisco ASIG.
1149 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1151 * gst/rtsp-server/rtsp-client.c:
1152 rtsp-client: Use watch_context before unref
1153 Move the usage of priv->watch_context to beginning of function
1154 gst_rtsp_client_finalize. Instead of use it after
1155 g_main_context_unref (priv->watch_context).
1157 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1159 * gst/rtsp-server/rtsp-stream.c:
1160 rtsp-stream: fix deadlock on transport removal
1161 We cannot take the RTSPStream lock while holding a transport backlog
1162 lock, as remove_transport may be called externally, which will
1163 take first the RTSPStream lock then the transport backlog lock.
1165 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1167 * gst/rtsp-server/rtsp-server-internal.h:
1168 * gst/rtsp-server/rtsp-stream-transport.c:
1169 * gst/rtsp-server/rtsp-stream.c:
1170 rtsp-stream: clear backlog when removing transport
1171 This ensures we don't end up calling any of transports' callbacks
1172 with a potentially unreffed user_data (in practice, a client that
1173 may have been removed)
1175 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1177 * gst/rtsp-server/rtsp-stream.c:
1178 rtsp-stream: marshal calls to send_tcp_message to a single thread
1179 In order to address the race condition pointed out at
1180 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1181 we get rid of the send thread pool, and instead spawn and manage
1182 a single thread to pull samples from app sinks and add them to
1183 the transport's backlogs.
1184 Additionally, we now also always go through the backlogs in order
1185 to simplify the logic.
1187 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1189 * gst/rtsp-server/rtsp-server-internal.h:
1190 * gst/rtsp-server/rtsp-stream-transport.c:
1191 * gst/rtsp-server/rtsp-stream.c:
1192 rtsp-stream: properly protect TCP backlog access
1194 We cannot hold stream->lock while pushing data, but need
1195 to consistently check the state of the backlog both from
1196 the send_tcp_message function and the on_message_sent function,
1197 which may or may not be called from the same thread.
1198 This commit introduces internal API to allow for potentially
1199 recursive locking of transport streams, addressing a race
1200 condition where the RTSP stream could push items out of order
1201 when popping them from the backlog.
1203 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1205 * gst/rtsp-server/rtsp-media.c:
1206 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1207 It's taken ownership of by the media, and returned with `transfer none`
1208 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1209 first then any bindings will wrongly take ownership of the pipeline once
1210 it arrives in bindings code.
1212 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1214 * examples/test-onvif-client.c:
1215 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1217 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1219 * gst/rtsp-server/rtsp-media.c:
1220 rtsp-media: fix default latency
1222 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1224 * gst/rtsp-server/rtsp-client.c:
1225 rtsp-client: make closing more thread safe
1226 + Take the watch lock prior to using priv->watch
1227 + Flush both the watch and connection before closing / unreffing
1228 gst_rtsp_connection_close() is not threadsafe on its own, this is
1229 a workaround at the client level, where we control both the watch
1232 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1234 * gst/rtsp-server/rtsp-latency-bin.c:
1235 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1238 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1239 `your_type_get_instance_private()` function instead
1242 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1244 * gst/rtsp-server/rtsp-client.c:
1245 * tests/check/gst/rtspserver.c:
1246 rtsp-client: add property post-session-timeout
1247 This is a TCP connection timeout for client connections, in seconds.
1248 If a positive value is set for this property, the client connection
1249 will be kept alive for this amount of seconds after the last session
1250 timeout. For negative values of this property the connection timeout
1251 handling is delegated to the system (just as it was before).
1254 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1256 * gst/rtsp-server/rtsp-stream.c:
1257 rtsp-stream: check for NULL transports prior to ref'ing
1259 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1261 * gst/rtsp-server/rtsp-server-internal.h:
1262 * gst/rtsp-server/rtsp-stream-transport.c:
1263 * gst/rtsp-server/rtsp-stream.c:
1264 rtsp-stream: fix checking of TCP backpressure
1265 The internal index of our appsinks, while it can be used to
1266 determine whether a message is RTP or RTCP, is not necessarily
1267 the same as the interleaved channel. Let the stream-transport
1268 determine the channel to check backpressure for, the same way
1269 it determines the channel according to whether it is sending
1272 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1274 * gst/rtsp-server/rtsp-session.c:
1275 rtsp-session: Butcher the file to please gst-indent in the CI
1276 This should be reverted once the CI has an updated gst-indent.
1278 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1280 * gst/rtsp-server/rtsp-session.c:
1281 * gst/rtsp-server/rtsp-session.h:
1282 * gst/rtsp-sink/gstrtspclientsink.c:
1283 * gst/rtsp-sink/gstrtspclientsink.h:
1284 rtsp-session & client: Remove deprecated GTimeVal
1285 GTimeVal won't work past 2038
1287 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1289 * gst/rtsp-server/rtsp-auth.c:
1290 rtsp-auth: fix default token leak
1292 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1294 * gst/rtsp-sink/gstrtspclientsink.c:
1295 gstrtspclientsink: unref transports when closing bin
1298 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1300 * gst/rtsp-server/rtsp-media.c:
1301 rtsp-media: Force seek when flush flag is set
1302 The commit "rtsp-client: define all seek accuracy flags from
1303 setup_play_mode" changed the behaviour of when doing a seek.
1304 Before that commit, having the flush flag set would result in a seek
1306 Even if no seek was needed. One reason to force seek is to flush old buffers
1307 created in Describe requests.
1308 Thus adding force seek also for flush flag will result in play request
1311 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1313 * gst/rtsp-server/rtsp-client.c:
1314 rtsp-client: Revitalize dead code
1315 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1318 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1320 * gst/rtsp-server/rtsp-sdp.c:
1321 rtsp-sdp: Don't try to use non-initialized values
1322 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1323 returns TRUE. Also avoid the whole clock signalling block if we're not
1324 dealing with senders.
1329 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1331 * gst/rtsp-server/rtsp-stream-transport.c:
1332 * gst/rtsp-server/rtsp-stream.c:
1333 * tests/check/gst/stream.c:
1334 rtsp-stream: Removing invalid transports returns false
1335 When removing transports an assertion was that the transports passed in
1336 for removal are present in the list, however that can't be assumed.
1337 As an example if a transport was removed from a thread running
1338 send_tcp_message, the main thread can try to remove the same transport
1339 again if it gets a handle_pause_request. This will not effect the
1340 transport list but it will effect n_tcp_transports as it will be
1341 decrement and then have the wrong value.
1343 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1345 * tests/check/gst/client.c:
1346 client test: add scale and speed negative tests
1347 Negative tests for scale and speed should be done as well, verify that
1348 the response code is "400 Bad request" when a bad request is done.
1350 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1352 * gst/rtsp-server/rtsp-auth.c:
1353 * gst/rtsp-server/rtsp-client.c:
1354 * gst/rtsp-server/rtsp-media-factory.c:
1355 * gst/rtsp-server/rtsp-media.c:
1356 * gst/rtsp-server/rtsp-server.c:
1357 * gst/rtsp-server/rtsp-session-pool.c:
1358 * gst/rtsp-server/rtsp-stream.c:
1359 * gst/rtsp-sink/gstrtspclientsink.c:
1360 Don't pass default GLib marshallers for signals
1361 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1362 actually internally optimize the signal (if the marshaller is available
1363 in GLib itself) by also setting the valist marshaller. This makes the
1364 signal emission a bit more performant than the regular marshalling,
1365 which still needs to box into `GValue` and call libffi in case of a
1367 Note that for custom marshallers, one would use
1368 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1370 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1372 * gst/rtsp-server/rtsp-mount-points.c:
1373 GstRTSPMountPoints: Remove any existing factory before adding a new one
1374 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1375 previous mount point will be freed" which was true when it was
1376 implemented using a GHashTable. But in 2012 it got rewrote using a
1377 GSequence and since then it could have 2 factories for the same path.
1378 Which one gets used is random, depending on the sorting order of 2
1381 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1383 * gst/rtsp-server/rtsp-client.c:
1384 * gst/rtsp-server/rtsp-server-internal.h:
1385 * gst/rtsp-server/rtsp-stream-transport.c:
1386 * gst/rtsp-server/rtsp-stream-transport.h:
1387 * gst/rtsp-server/rtsp-stream.c:
1388 stream: refactor TCP backpressure handling
1389 The previous implementation stopped sending TCP messages to
1390 all clients when a single one stopped consuming them, which
1391 obviously created problems for shared media.
1392 Instead, we now manage a backlog in stream-transport, and slow
1393 clients are removed once this backlog exceeds a maximum duration,
1394 currently hardcoded.
1397 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1400 meson: build gir even when cross-compiling if introspection was enabled explicitly
1401 This can be made to work in certain circumstances when
1402 cross-compiling, so default to not building g-i stuff
1403 when cross-compiling, but allow it if introspection was
1404 enabled explicitly via -Dintrospection=enabled.
1405 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1407 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1409 * gst/rtsp-server/rtsp-session.c:
1410 rtsp-session: clean up comment extra-timeout
1412 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1414 * gst/rtsp-server/rtsp-client.c:
1415 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1416 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1417 from the RTSP context.
1420 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1422 * gst/rtsp-server/rtsp-client.c:
1423 * gst/rtsp-server/rtsp-media.c:
1424 * gst/rtsp-server/rtsp-media.h:
1425 rtsp-client: Lock shared media
1426 For shared media we got race conditions. Concurrently rtsp clients might
1427 suspend or unsuspend the shared media and thus change the state without
1428 the clients expecting that.
1429 By introducing a lock that can be taken by callers such as rtsp_client
1430 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1431 to handle the media sequentially thus allowing one client to finish its
1432 rtsp call before another client calls on the same media.
1433 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1436 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1438 * gst/rtsp-server/rtsp-session.c:
1439 rtsp-session: add property extra-timeout
1440 Extra time to add to the timeout, in seconds. This only
1441 affects the time until a session is considered timed out
1442 and is not signalled in the RTSP request responses.
1443 Only the value of the timeout property is signalled in the
1446 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1448 * gst/rtsp-server/rtsp-stream.c:
1449 rtsp-stream : fix race condition in send_tcp_message
1450 If one thread is inside the send_tcp_message function and are done
1451 sending rtp or rtcp messages so the n_outstanding variable is zero
1452 however have not exit the loop sending the messages. While sending its
1453 messages, transports have been added or removed to the transport list,
1454 so the cache should be updated. If now an additional thread comes to
1455 the function send_tcp_message and trying to send rtp messages it will
1456 first destroy the rtp cache that is still being iterated trough by the
1460 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1469 * examples/.gitignore:
1470 * examples/Makefile.am:
1472 * gst/rtsp-server/.gitignore:
1473 * gst/rtsp-server/Makefile.am:
1474 * gst/rtsp-sink/Makefile.am:
1475 * pkgconfig/.gitignore:
1476 * pkgconfig/Makefile.am:
1478 * tests/Makefile.am:
1479 * tests/check/Makefile.am:
1480 Remove autotools build
1482 Maybe we can now use the meson pkgconfig module
1483 for .pc files? (Does it support uninstalled now?)
1485 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1487 * tests/check/gst/client.c:
1488 client: fix test mem leak in attach_rate_tweaking_probe
1490 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1492 * tests/check/gst/media.c:
1493 media: remove memleak in test test_media_seek
1495 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1497 * tests/check/gst/rtspserver.c:
1498 rtspserver: Remove memleak in test test_double_play
1500 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1502 * gst/rtsp-server/rtsp-media.c:
1503 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1505 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1507 * gst/rtsp-server/rtsp-media.c:
1508 * tests/check/gst/rtspserver.c:
1509 rtsp-media: Unblock all streams
1510 When unsuspending and going to PLAYING, unblock all streams instead of
1511 only those that are linked (the linked streams are the ones for which
1512 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1513 pushing buffers on unlinked streams.
1514 This change is because playback using single-threaded demuxers like
1515 matroska-demux could be blocked if SETUP was not called for all media.
1516 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1517 gstflvdemux, qtdemux, and matroska-demux) will handle
1518 GST_FLOW_NOT_LINKED automatically.
1521 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1523 * gst/rtsp-server/rtsp-media.c:
1524 * tests/check/gst/rtspserver.c:
1525 rtsp-media: Wait on async when needed.
1526 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1527 In the unit test the pause from adjust_play_mode will cause a preroll
1528 and after that async-done will be produced.
1529 Without this patch there are no one consuming this async-done and when
1530 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1531 wait for async-done. But then it wrongly find the async-done prodused by
1532 adjus_play_mode and continue executing without waiting for the preroll
1535 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1537 * gst/rtsp-server/rtsp-client.c:
1538 rtsp-client: RTP Info when completed_sender
1539 Change condition that should be fulfilled regarding RTPInfo.
1540 Replace !gst_rtsp_media_is_receive_only with
1541 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1542 for a sender pipeline that is complete. Only then a RTPInfo should
1544 gst_rtsp_media_is_receive_only gives different answears depending on
1546 If Describe is called wth URL+options for backchannel SDP will give only
1547 audio and only backchannel a=sendonly
1548 If Describe is called on URL+options that gives both audio and video
1549 direction from server to client, pipelines are created. Thus
1550 receive_only will return false, even though Setup only would setup
1552 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1553 streams are complete.
1555 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1557 * gst/rtsp-server/rtsp-client.c:
1558 * tests/check/gst/client.c:
1559 rtsp-client: RTP Info exists conditionally in PLAY
1560 If RTP Info is missing and it is not a receiver only, eg. audio
1561 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1562 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1563 Since 1.14 there is audio backchannel support. Thus RTP-info is
1564 conditional now. When audio backchannel only mode, there is no RTP-info.
1567 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1569 * examples/test-onvif-client.c:
1570 test-onvif-client: remove unused query
1572 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1574 * gst/rtsp-server/rtsp-client.c:
1575 rtsp-client: RTP Info must exist in PLAY response
1576 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1579 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1581 * examples/test-onvif-client.c:
1582 test-onvif-client: perform accurate seeks
1583 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1584 Also, modify how we compute the position: position queries in
1585 PAUSED mode fail to account for the newly-prerolled frame, leading
1586 to frame skips when performing seeks in that state. Instead,
1587 compute the current position from the last sample.
1589 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1591 * gst/rtsp-server/rtsp-client.c:
1592 * gst/rtsp-server/rtsp-media.c:
1593 * gst/rtsp-server/rtsp-media.h:
1594 * tests/check/gst/rtspserver.c:
1595 Use complete streams for scale and speed.
1596 Without this patch it's always stream0 that is used to get segment event
1597 that is used to set scale and speed. This even if client not doing SETUP
1598 for stream0. At least in suspend mode reset this not working since then
1599 it's just random if send_rtp_sink have got any segment event. There are
1600 no check if send_rtp_sink for stream0 got any data before media is
1601 prerolled after PLAY request.
1603 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1605 * examples/test-onvif-server.c:
1606 * examples/test-onvif-server.h:
1607 examples/onvif-server: fix werror build with clang
1608 ../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]
1609 self->incoming_segment->format, self->incoming_segment->flags,
1610 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1611 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1612 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1614 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1615 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1617 <scratch space>:77:1: note: expanded from here
1620 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1621 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1623 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1624 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1626 <scratch space>:9:1: note: expanded from here
1629 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1630 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1631 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1633 <scratch space>:12:1: note: expanded from here
1637 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1640 meson: Don't generate doc cache when no plugins are enabled
1641 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1643 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1645 * examples/test-onvif-client.c:
1646 test-onvif-client: stdin is not defined in MSVC
1648 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1650 * gst/rtsp-server/rtsp-media.c:
1651 rtsp-media: add missing Since tag
1653 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1655 * examples/test-onvif-client.c:
1656 test-onvif-client: STDIN_FILENO is not portable
1657 If not defined, define it to _fileno(stdin) on Windows, 0
1660 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1662 * examples/test-onvif-server.c:
1663 test-onvif-server: downgrade logging
1665 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1667 * examples/meson.build:
1668 * examples/test-onvif-client.c:
1669 * examples/test-onvif-server.c:
1670 examples: add ONVIF client / server example
1672 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1674 * gst/rtsp-server/rtsp-client.c:
1675 * gst/rtsp-server/rtsp-media.c:
1676 rtsp-client: define all seek accuracy flags from setup_play_mode
1677 We then pass those to adjust_play_mode, which needs to operate
1678 on the "final" seek flags, as previously the code in rtsp-media
1679 was assuming that accuracy seek flags (accurate / key_unit) should
1680 not be set if the flags passed to the seek method were already set.
1682 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1684 * gst/rtsp-server/rtsp-media-factory-uri.c:
1685 * gst/rtsp-server/rtsp-media.c:
1686 rtsp-media: Try to get dynamic payloaders by name from their bin first
1687 First try "pay", then "pay_%s" (where %s == pad name). And only then
1688 fall back to the code that simply takes the first payloader that is
1690 The current code usually works (but is racy) because it will always take
1691 the payloader that was last added (due to g_list_prepend() when adding
1692 elements) in pad-added and that's usually the correct one. But if a new
1693 payloader is added between pad-added and us trying to get it, we would
1694 get the wrong payloader.
1696 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1698 * tests/check/gst/client.c:
1699 client test: expect any port in transport
1700 setup_multicast_client sets a 5000-5010 range for the client
1701 ports, it is incorrect to expect the transport to always use
1705 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1707 * tests/check/gst/onvif.c:
1708 onvif tests: use g_cond_wait() correctly
1709 g_cond_wait() has to be called in a loop until required conditions
1713 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1715 * gst/rtsp-server/rtsp-stream.c:
1716 rtsp-stream: Not wait on receiver streams when pre-rolling
1717 Without this patch there are problem pre-rolling when using audio back
1719 Without this patch a probe will be created for all streams including
1720 the stream for audio backchannel. To pre-roll all this pads have to
1721 receive data. Since the stream for audio backchannel is a receiver this
1723 The solution is to never create any probes for streams that are for
1724 incomming data and instead set them as blocking already from beginning.
1726 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1728 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1729 * gst/rtsp-server/rtsp-onvif-media.c:
1730 onvif-media: fix "void function returning a value" compiler warning
1732 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1734 * gst/rtsp-server/rtsp-media.c:
1735 rtsp-media: make sure streams are blocked when sending seek
1736 The recent ONVIF work exposed a race condition when dealing with
1737 multiple streams: one of the sinks may preroll before other streams
1738 have started flushing. This led to the pipeline posting async-done
1739 prematurely, when some streams were actually still in the middle
1740 of performing a flushing seek. The newly-added code looks up a
1741 sticky segment event on the first stream in order to respond to
1742 the PLAY request with accurate Scale and Speed headers. In the
1743 failure condition, the first stream was flushing, and thus had
1744 no sticky segment event, leading to the PLAY request failing,
1745 and in turn the test.
1747 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
1750 * gst/rtsp-server/rtsp-media-factory-uri.h:
1753 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1755 * gst/rtsp-server/rtsp-client.c:
1756 * gst/rtsp-server/rtsp-client.h:
1757 * gst/rtsp-server/rtsp-media.c:
1758 * gst/rtsp-server/rtsp-media.h:
1759 * gst/rtsp-server/rtsp-onvif-client.c:
1760 * gst/rtsp-server/rtsp-onvif-client.h:
1761 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1762 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1763 * gst/rtsp-server/rtsp-onvif-media.c:
1764 * gst/rtsp-server/rtsp-onvif-server.h:
1765 * gst/rtsp-server/rtsp-stream.c:
1766 * gst/rtsp-server/rtsp-stream.h:
1767 * tests/check/gst/media.c:
1768 * tests/check/gst/onvif.c:
1769 * tests/check/meson.build:
1770 onvif: Implement and test the Streaming Specification
1771 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
1773 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1775 * gst/rtsp-server/rtsp-client.c:
1776 * gst/rtsp-server/rtsp-client.h:
1777 rtsp-client: add gst_rtsp_client_get_stream_transport()
1778 This will be used in the onvif tests in order to validate the
1779 data transmitted over TCP: for streaming to continue after a
1780 data message has been provided to client->send_func, the client
1781 is responsible for marking the message as sent on the relevant
1784 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1786 * gst/rtsp-server/rtsp-client.c:
1787 client: Scale implies TRICK_MODE
1789 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1791 * gst/rtsp-server/rtsp-client.c:
1792 client: compare booleans, not pointers to them
1794 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
1796 * gst/rtsp-server/rtsp-media.c:
1797 * gst/rtsp-server/rtsp-stream.c:
1798 * tests/check/gst/media.c:
1799 Reverse playback support
1800 GStreamer plays segment from stop to start when doing reverse playback.
1801 RTSP implies that media should be played from start of Range header to
1802 its stop. Hence we swap start and stop times before passing them to
1804 Also make gst_rtsp_stream_query_stop always return value that can be
1805 used as stop time of Range header.
1807 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
1809 * gst/rtsp-server/rtsp-client.c:
1810 * gst/rtsp-server/rtsp-media.c:
1811 * gst/rtsp-server/rtsp-media.h:
1812 * tests/check/gst/client.c:
1813 rtsp-client: add support for Scale and Speed header
1814 Add support for the RTSP Scale and Speed headers by setting the rate in
1815 the seek to (scale*speed). We then check the resulting segment for rate
1816 and applied rate, and use them as values for the Speed and Scale headers
1818 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1820 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
1822 * gst/rtsp-server/rtsp-client.c:
1823 * gst/rtsp-server/rtsp-client.h:
1824 rtsp-client: allow sub classes to adjust the seek
1825 Adds a new virtual function, adjust_play_mode(), that allows
1826 sub classes to adjust the seek done on the media. The sub class can
1827 modify the values of the the seek flags and the rate.
1828 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1830 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
1832 * gst/rtsp-server/rtsp-media.c:
1833 * gst/rtsp-server/rtsp-media.h:
1834 * gst/rtsp-server/rtsp-stream.c:
1835 * gst/rtsp-server/rtsp-stream.h:
1836 * tests/check/gst/media.c:
1837 rtsp-media: allow specifying rate when seeking
1838 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
1839 caller to specify the rate for the seek. Also added functions in
1840 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
1841 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1843 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
1847 meson: Bump minimal GLib version to 2.44
1848 This means we can use some newer features and get rid of some
1849 boilerplate code using the G_DECLARE_* macros.
1850 As discussed on IRC, 2.44 is old enough by now to start depending on it.
1852 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1854 * docs/libs/.gitignore:
1855 * docs/libs/Makefile.am:
1856 * docs/libs/gst-rtsp-server-docs.sgml:
1857 * docs/libs/gst-rtsp-server-sections.txt:
1858 * docs/libs/gst-rtsp-server.types:
1859 docs: remove obsolete gtk-doc related files
1861 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1863 * gst/rtsp-sink/gstrtspclientsink.c:
1864 doc: remove xml from comments
1866 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
1868 * docs/gst_plugins_cache.json:
1870 docs: Stop building the doc cache by default
1871 And update the cache
1872 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
1874 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
1876 * docs/gst_plugins_cache.json:
1877 docs: Update plugins documentation cache
1879 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
1882 * gst/rtsp-server/rtsp-context.c:
1883 * gst/rtsp-server/rtsp-session-pool.c:
1884 doc: Fix some docstrings
1886 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
1892 * docs/gst_plugins_cache.json:
1895 * docs/plugin-index.md:
1896 * docs/plugin-sitemap.txt:
1899 * docs/version.entities.in:
1900 * gst/rtsp-server/meson.build:
1901 * gst/rtsp-sink/meson.build:
1903 * meson_options.txt:
1904 docs: Port to hotdoc
1906 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1908 * gst/rtsp-server/rtsp-auth.c:
1909 * gst/rtsp-server/rtsp-client.h:
1910 rtsp-server: Fix various Since markers
1912 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1914 * gst/rtsp-server/rtsp-media.c:
1915 * gst/rtsp-server/rtsp-sdp.c:
1916 * gst/rtsp-server/rtsp-session-media.c:
1917 * gst/rtsp-server/rtsp-stream.c:
1918 rtsp-server: Add various Since: 1.14 markers
1920 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1922 * gst/rtsp-server/rtsp-media-factory.c:
1923 * gst/rtsp-server/rtsp-media.c:
1924 * gst/rtsp-server/rtsp-stream-transport.c:
1925 * gst/rtsp-server/rtsp-stream.c:
1926 rtsp-server: Add various missing Since: 1.16 markers
1928 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
1930 * gst/rtsp-sink/gstrtspclientsink.c:
1931 rtspclientsink: Set async-handling=false for the internal bins
1932 Without this we can easily run into a race condition with async state changes:
1933 - the pipeline is doing an async state change
1934 - we set the internal bins to PLAYING but that's ignored because an
1935 async state change is currently pending
1936 - the async state change finishes but does not change the state of the
1937 internal bins because of locked_state==TRUE
1938 - the internal bins stay in PAUSED forever
1940 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1942 * gst/rtsp-sink/gstrtspclientsink.c:
1943 rtspclientsink: Use write_messages() API to send buffer lists in one go
1944 And to write messages with multiple memories also via writev().
1946 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
1948 * gst/rtsp-server/rtsp-client.c:
1949 * gst/rtsp-server/rtsp-client.h:
1950 * gst/rtsp-server/rtsp-server-object.h:
1951 * gst/rtsp-server/rtsp-server.c:
1952 rtsp-client: Handle Content-Length limitation
1953 Add functionality to limit the Content-Length.
1954 API addition, Enhancement.
1955 Define an appropriate request size limit and reject requests
1956 exceeding the limit with response status 413 Request Entity Too Large
1959 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1966 === release 1.16.0 ===
1968 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1974 * gst-rtsp-server.doap:
1978 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
1980 * gst/rtsp-sink/gstrtspclientsink.c:
1981 rtspclientsink: Notify the stream transport about each written message
1982 Otherwise it will never try to send us the next one: it tries to keep
1983 exactly one message in-flight all the time.
1984 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
1985 in the client sink we always write data out synchronously.
1987 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
1989 * gst/rtsp-server/rtsp-stream.c:
1990 rtsp_server: Free thread pool before clean transport cache
1991 If not waiting for free thread pool before clean transport caches, there
1992 can be a crash if a thread is executing in transport list loop in
1993 function send_tcp_message.
1994 Also add a check if priv->send_pool in on_message_sent to avoid that a
1995 new thread is pushed during wait of free thread pool. This is possible
1996 since when waiting for free thread pool mutex have to be unlocked.
1998 === release 1.15.90 ===
2000 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
2006 * gst-rtsp-server.doap:
2010 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
2012 * gst/rtsp-server/rtsp-stream.c:
2013 rtsp-stream: Add support for GCM (RFC 7714)
2016 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2018 * gst/rtsp-server/rtsp-session-pool.c:
2019 session pool: fix missing klass-> in klass->create_session
2021 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2024 g-i: pass --quiet to g-ir-scanner
2025 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2026 that we get even if everything works just fine.
2027 We still get g-ir-scanner warnings and compiler warnings if
2028 we pass this option.
2030 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2033 g-i: silence 'nested extern' compiler warnings when building scanner binary
2034 We need a nested extern in our init section for the scanner binary
2035 so we can call gst_init to make sure GStreamer types are initialised
2036 (they are not all lazy init via get_type functions, but some are in
2037 exported variables). There doesn't seem to be any other mechanism to
2038 achieve this, so just remove that warning, it's not important at all.
2040 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2043 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2045 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2047 * gst/rtsp-server/rtsp-media.c:
2048 * tests/check/gst/media.c:
2049 rtsp-media: Handle set state when preparing.
2050 Handle the situation when a call to gst_rtsp_media_set_state is done
2051 when media status is preparing.
2052 Also add unit test for this scenario.
2053 The unit test simulate on a media level when two clients share a (live)
2055 Both clients have done SETUP and got responses. Now client 1 is doing
2056 play and client 2 is just closing the connection.
2057 Then without patch there are a problem when
2058 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2059 And client2 is doing closing connection we can end up in a call
2060 to gst_rtsp_media_set_state when
2061 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2062 shut down media is jumped over .
2063 With this patch and this scenario we wait until
2064 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2065 execute after that and now we will execute the logic for
2068 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2076 === release 1.15.2 ===
2078 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2084 * gst-rtsp-server.doap:
2088 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2090 * gst/rtsp-server/rtsp-media.c:
2091 * tests/check/gst/client.c:
2092 rtsp-media: Fix multicast use case with common media
2101 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2103 * gst/rtsp-server/rtsp-client.c:
2104 * gst/rtsp-server/rtsp-stream.c:
2105 * gst/rtsp-server/rtsp-stream.h:
2106 rtsp-server: remove recursive behavior
2107 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2109 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2111 * gst/rtsp-server/rtsp-client.c:
2112 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2113 And route all messages through the send_func if no send_messages_func
2115 We otherwise break backwards compatibility.
2117 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2119 * docs/libs/gst-rtsp-server-sections.txt:
2120 * gst/rtsp-server/rtsp-client.c:
2121 * gst/rtsp-server/rtsp-client.h:
2122 * gst/rtsp-server/rtsp-stream.c:
2123 rtsp-client: Add support for sending buffer lists directly
2124 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2126 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2128 * docs/libs/gst-rtsp-server-sections.txt:
2129 * gst/rtsp-server/rtsp-client.c:
2130 * gst/rtsp-server/rtsp-media.c:
2131 * gst/rtsp-server/rtsp-stream-transport.c:
2132 * gst/rtsp-server/rtsp-stream-transport.h:
2133 * gst/rtsp-server/rtsp-stream.c:
2134 * gst/rtsp-sink/gstrtspclientsink.c:
2135 rtsp-server: Add support for buffer lists
2136 This adds new functions for passing buffer lists through the different
2137 layers without breaking API/ABI, and enables the appsink to actually
2138 provide buffer lists.
2139 This should already reduce CPU usage and potentially context switches a
2140 bit by passing a whole buffer list from the appsink instead of
2141 individual buffers. As a next step it would be necessary to
2142 a) Add support for a vector of data for the GstRTSPMessage body
2143 b) Add support for sending multiple messages at once to the
2144 GstRTSPWatch and let it be handled internally
2145 c) Adding API to GOutputStream that works like writev()
2146 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2148 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2150 * gst/rtsp-server/rtsp-client.c:
2151 client: Fix crash in close handler
2152 The close handler could trigger a crash because it invalidated the
2153 watch_context while still leaving a source attached to it which would be
2154 cleaned up at a later point.
2156 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2158 * gst/rtsp-server/rtsp-stream.c:
2159 rtsp-stream: Use cached address when allocating sockets
2160 If an address/port was previously decided upon (ex: multicast in the
2161 SDP), then use that instead of re-creating another one
2162 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2164 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2166 * gst/rtsp-server/rtsp-media.c:
2167 rtsp-media: Fix race codition in finish_unprepare
2168 The previous fix for race condition around finish_unprepare where the
2169 function could be called twice assumed that the status wouldn't change
2170 during execution of the function. This assumption is incorrect as the
2171 state may change, for example if an error message arrives from the
2173 Instead a flag keeping track on whether the finish_unprepare function
2174 is currently executing is introduced and checked.
2175 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2177 === release 1.15.1 ===
2179 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2185 * gst-rtsp-server.doap:
2189 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2191 * gst/rtsp-server/rtsp-stream.c:
2192 Add source elements to the pipeline before activation
2193 In plug_src we changed the element state before adding it to
2194 the owner container. This prevented the pipeline from intercepting
2195 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2196 to assign a custom task pool.
2197 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2199 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2202 Automatic update of common submodule
2203 From ed78bee to 59cb678
2205 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2207 * examples/test-appsrc.c:
2208 examples: test-appsrc: fix coding style error
2210 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2212 * examples/test-appsrc.c:
2213 examples: test-appsrc: fix buffer leak
2215 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2217 * gst/rtsp-server/rtsp-media.c:
2218 rtsp-media: Update priv->blocked when linked streams are unblocked.
2219 Media is considered to be blocked when all streams that belong to
2220 that media are blocked.
2221 This patch solves the problem of inconsistent updates of
2222 priv->blocked that are not synchronized with the media state.
2224 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2226 * gst/rtsp-server/rtsp-media.c:
2227 rtsp-media: Don't block streams before seeking
2228 Before the seek operation is performed on media, it's required that
2229 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2230 At this stage, all transport parts (transport sinks) have been successfully
2231 added to the pipeline and there is no need for blocking the streams.
2233 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2235 * tests/check/gst/rtspserver.c:
2236 tests: rtspserver: Add shared media test case for TCP
2238 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2240 * gst/rtsp-server/rtsp-stream.c:
2241 rtsp-stream: Use seqnum-offset for rtpinfo
2242 The sequence number in the rtpinfo is supposed to be the first RTP
2243 sequence number. The "seqnum" property on a payloader is supposed to be
2244 the number from the last processed RTP packet. The sequence number for
2245 payloaders that inherit gstrtpbasepayload will not be correct in case of
2246 buffer lists. In order to fix the seqnum property on the payloaders
2247 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2248 "seqnum-offset" from the "stats" property contains the value of the
2249 very first RTP packet in a stream. The server will, however, try to look
2250 at the last simple in the sink element and only use properties on the
2251 payloader in case there no sink elements yet, and by looking at the last
2252 sample of the sink gives the server full control of which RTP packet it
2253 looks at. If the payloader does not have the "stats" property, "seqnum"
2254 is still used since "seqnum-offset" is only present in as part of
2255 "stats" and this is still an issue not solved with this patch.
2256 Needed for gst-plugins-base!17
2258 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2260 * gst/rtsp-server/rtsp-stream.c:
2261 rtsp-stream: Plug memory leak
2262 Attaching a GSource to a context will increase the refcount. The idle
2263 source will never be free'd since the initial reference is never
2266 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2269 Add Gitlab CI configuration
2270 This commit adds a .gitlab-ci.yml file, which uses a feature
2271 to fetch the config from a centralized repository. The intent is
2272 to have all the gstreamer modules use the same configuration.
2273 The configuration is currently hosted at the gst-ci repository
2274 under the gitlab/ci_template.yml path.
2275 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2277 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2280 * gst-rtsp-server.doap:
2281 Update git locations to gitlab
2283 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2285 * gst/rtsp-server/meson.build:
2286 meson: add new onvif types
2288 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2290 * gst/rtsp-server/meson.build:
2291 Add ONVIF subclass headers to the installed headers in meson.build too
2293 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2295 * gst/rtsp-server/rtsp-server-object.h:
2296 * gst/rtsp-server/rtsp-server.h:
2297 rtsp-server: Declare GstRTSPServer struct before anything else
2298 It's needed by all kinds of other headers, including the ones that are
2299 required for defining the GstRTSPServer struct itself and its API.
2301 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2303 * gst/rtsp-server/rtsp-onvif-client.h:
2304 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2305 * gst/rtsp-server/rtsp-onvif-media.h:
2306 * gst/rtsp-server/rtsp-onvif-server.h:
2307 Mark all ONVIF-specific subclasses as Since 1.14
2309 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2311 * gst/rtsp-server/Makefile.am:
2312 * gst/rtsp-server/meson.build:
2313 * gst/rtsp-server/rtsp-context.h:
2314 * gst/rtsp-server/rtsp-onvif-server.c:
2315 * gst/rtsp-server/rtsp-onvif-server.h:
2316 * gst/rtsp-server/rtsp-server-object.h:
2317 * gst/rtsp-server/rtsp-server-prelude.h:
2318 * gst/rtsp-server/rtsp-server.c:
2319 * gst/rtsp-server/rtsp-server.h:
2320 * gst/rtsp-server/rtsp-session.h:
2321 Include ONVIF types from single-include rtsp-server.h
2322 ... by actually making it a single-include header and moving everything
2323 related to the GstRTSPServer type to rtsp-server-object.h instead.
2324 Otherwise there are too many circular includes.
2325 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2327 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2329 * gst/rtsp-server/rtsp-client.c:
2330 * gst/rtsp-server/rtsp-latency-bin.c:
2331 * gst/rtsp-server/rtsp-stream.c:
2332 * gst/rtsp-server/rtsp-stream.h:
2333 rtsp-stream: use idle source in on_message_sent
2334 When the underlying layers are running on_message_sent, this sometimes
2335 causes the underlying layer to send more data, which will cause the
2336 underlying layer to run callback on_message_sent again. This can go on
2338 To break this chain, we introduce an idle source that takes care of
2339 sending data if there are more to send when running callback
2340 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2342 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2344 * gst/rtsp-server/rtsp-client.c:
2345 rtsp-client: Remove timeout GSource on cleanup
2346 Avoids ending up with races where a timeout would still be around
2347 *after* a client was gone. This could happen rather easily in
2348 RTSP-over-HTTP mode on a local connection, where each RTSP message
2349 would be sent as a different HTTP connection with the same tunnelid.
2350 If not properly removed, that timeout would then try to free again
2351 a client (and its contents).
2353 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2355 * gst/rtsp-server/Makefile.am:
2356 autotools: fix distcheck
2358 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2360 * gst/rtsp-server/Makefile.am:
2361 * gst/rtsp-server/meson.build:
2362 * gst/rtsp-server/rtsp-latency-bin.c:
2363 * gst/rtsp-server/rtsp-latency-bin.h:
2364 * gst/rtsp-server/rtsp-onvif-media.c:
2365 onvif: encapsulate onvif part into a bin
2366 ...and thus do not let onvif affect pipelines latency
2367 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2369 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2371 * tests/check/gst/client.c:
2372 tests: client: Avoid bind() failures in tests
2373 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2375 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2377 * gst/rtsp-server/rtsp-media-factory.c:
2378 * gst/rtsp-server/rtsp-media-factory.h:
2379 * gst/rtsp-server/rtsp-media.c:
2380 * gst/rtsp-server/rtsp-media.h:
2381 * gst/rtsp-server/rtsp-stream.c:
2382 * gst/rtsp-server/rtsp-stream.h:
2383 * tests/check/gst/client.c:
2384 * tests/check/gst/mediafactory.c:
2385 New property for socket binding to mcast addresses
2386 By default the multicast sockets are bound to INADDR_ANY,
2387 as it's not allowed to bind sockets to multicast addresses
2388 in Windows. This default behaviour can be changed by setting
2389 bind-mcast-address property on the media-factory object.
2390 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2392 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2395 * gst/rtsp-server/Makefile.am:
2396 * gst/rtsp-server/meson.build:
2397 * gst/rtsp-server/rtsp-address-pool.c:
2398 * gst/rtsp-server/rtsp-auth.c:
2399 * gst/rtsp-server/rtsp-client.c:
2400 * gst/rtsp-server/rtsp-context.c:
2401 * gst/rtsp-server/rtsp-media-factory-uri.c:
2402 * gst/rtsp-server/rtsp-media-factory.c:
2403 * gst/rtsp-server/rtsp-media.c:
2404 * gst/rtsp-server/rtsp-mount-points.c:
2405 * gst/rtsp-server/rtsp-params.c:
2406 * gst/rtsp-server/rtsp-permissions.c:
2407 * gst/rtsp-server/rtsp-sdp.c:
2408 * gst/rtsp-server/rtsp-server-prelude.h:
2409 * gst/rtsp-server/rtsp-server.c:
2410 * gst/rtsp-server/rtsp-session-media.c:
2411 * gst/rtsp-server/rtsp-session-pool.c:
2412 * gst/rtsp-server/rtsp-session.c:
2413 * gst/rtsp-server/rtsp-stream-transport.c:
2414 * gst/rtsp-server/rtsp-stream.c:
2415 * gst/rtsp-server/rtsp-thread-pool.c:
2416 * gst/rtsp-server/rtsp-token.c:
2418 libs: fix API export/import and 'inconsistent linkage' on MSVC
2419 Export rtsp-server library API in headers when we're building the
2420 library itself, otherwise import the API from the headers.
2421 This fixes linker warnings on Windows when building with MSVC.
2422 Fix up some missing config.h includes when building the lib which
2423 is needed to get the export api define from config.h
2424 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2426 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2428 * gst/rtsp-server/rtsp-media-factory.c:
2429 rtsp-media-factory: Add missing break statements
2430 This resulted in warnings/assertions whenever one accessed the
2431 max-mcast-ttl property.
2435 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2438 * meson_options.txt:
2439 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2441 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2444 * meson_options.txt:
2445 * tests/check/meson.build:
2446 meson: add option to disable build of rtspclientsink plugin
2448 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2450 * meson_options.txt:
2451 meson: re-arrange options
2453 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2456 * meson_options.txt:
2457 * tests/check/meson.build:
2458 * tests/meson.build:
2459 meson: Use feature option for tests option
2460 This was somehow missed the last time around.
2462 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2464 * gst/rtsp-server/meson.build:
2466 meson: Maintain macOS ABI through dylib versioning
2467 Requires Meson 0.48, but the feature will be ignored on older versions
2468 so it's safe to add it without bumping the requirement.
2470 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2472 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2474 * gst/rtsp-sink/meson.build:
2476 meson: add pkg-config file for the rtspclientsink plugin
2478 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2480 * gst/rtsp-server/rtsp-client.c:
2481 * tests/check/gst/client.c:
2482 rtsp-client: Avoid reuse of channel numbers for interleaved
2483 If a (strange) client would reuse interleaved channel numbers in
2484 multiple SETUP requests, we should not accept them. The channel
2485 numbers are used for looking up stream transports in the
2486 priv->transports hash table, and transports disappear from the table
2487 if channel numbers are reused.
2488 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2489 server to change the channel numbers suggested by the client.
2490 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2492 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2494 * tests/check/gst/client.c:
2495 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2496 Allow regex for matching transport header against expected pattern.
2497 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2499 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2501 * tests/check/meson.build:
2502 meson: There is no gstreamer-plugins-good-1.0.pc
2503 There is no installed version of that, only an uninstalled version.
2505 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2507 * gst/rtsp-server/rtsp-client.c:
2508 * tests/check/gst/stream.c:
2509 Fix indentation again
2511 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2513 * gst/rtsp-server/rtsp-client.c:
2514 * gst/rtsp-server/rtsp-stream.c:
2515 * gst/rtsp-server/rtsp-stream.h:
2516 * tests/check/gst/client.c:
2517 * tests/check/gst/stream.c:
2518 stream: Added a list of multicast client addresses
2519 When media is shared, the same media stream can be sent
2520 to multiple multicast groups. Currently, there is no API
2521 to retrieve multicast addresses from the stream.
2522 When calling gst_rtsp_stream_get_multicast_address() function,
2523 only the first multicast address is returned.
2524 With this patch, each multicast destination requested in SETUP
2525 will be stored in an internal list (call to
2526 gst_rtsp_stream_add_multicast_client_address()).
2527 The list of multicast groups requested by the clients can be
2528 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2529 There still exist some problems with the current implementation
2530 in the multicast case:
2531 1) The receiving part is currently only configured with
2532 regard to the first multicast client (see
2533 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2534 2) Secondly, of security reasons, some constraints should be
2535 put on the requested multicast destinations (see
2536 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2537 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2538 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2540 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2542 * gst/rtsp-server/rtsp-client.c:
2543 * gst/rtsp-server/rtsp-stream.c:
2544 * gst/rtsp-server/rtsp-stream.h:
2545 * tests/check/gst/client.c:
2546 stream: Choose the maximum ttl value provided by multicast clients
2547 The maximum ttl value provided so far by the multicast clients
2548 will be chosen and reported in the response to the current
2550 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2551 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2553 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2555 * gst/rtsp-server/rtsp-stream.c:
2556 * tests/check/gst/client.c:
2557 rtsp-stream: Don't require address pool in the transport specific case
2558 If "transport.client-settings" parameter is set to true, the client is
2559 allowed to specify destination, ports and ttl.
2560 There is no need for pre-configured address pool.
2561 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2562 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2564 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2566 * gst/rtsp-server/rtsp-client.c:
2567 * tests/check/gst/client.c:
2568 client: Don't reserve multicast address in the client setting case
2569 When two multicast clients request specific transport
2570 configurations, and "transport.client-settings" parameter is
2571 set to true, it's wrong to actually require that these two
2572 clients request the same multicast group.
2573 Removed test_client_multicast_invalid_transport_specific test
2574 cases as they wrongly require that the requested destination
2575 address is supposed to be present in the address pool, also in
2576 the case when "transport.client-settings" parameter is set to true.
2577 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2578 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2580 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2582 * gst/rtsp-server/rtsp-media-factory.c:
2583 * gst/rtsp-server/rtsp-media-factory.h:
2584 * gst/rtsp-server/rtsp-media.c:
2585 * gst/rtsp-server/rtsp-media.h:
2586 * gst/rtsp-server/rtsp-stream.c:
2587 * gst/rtsp-server/rtsp-stream.h:
2588 * tests/check/gst/mediafactory.c:
2589 Add new API for setting/getting maximum multicast ttl value
2590 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2591 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2593 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2595 * gst/rtsp-server/rtsp-stream.c:
2596 rtsp-stream: avoid duplicating the first multicast client
2597 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2598 clients were dynamically added and removed to the multicast
2599 udp sinks, as such we should no longer add a first client in
2600 set_multicast_socket_for_udpsink
2601 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2603 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2605 * gst/rtsp-server/rtsp-stream.c:
2606 Revert "rtsp-stream: avoid duplicating the first multicast client"
2607 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2608 Commits where accidentially squashed together
2610 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2612 * gst/rtsp-server/rtsp-client.c:
2613 * gst/rtsp-server/rtsp-media-factory.c:
2614 * gst/rtsp-server/rtsp-media-factory.h:
2615 * gst/rtsp-server/rtsp-media.c:
2616 * gst/rtsp-server/rtsp-media.h:
2617 * gst/rtsp-server/rtsp-stream.c:
2618 * gst/rtsp-server/rtsp-stream.h:
2619 * tests/check/gst/client.c:
2620 * tests/check/gst/mediafactory.c:
2621 Revert "Add new API for setting/getting maximum multicast ttl value"
2622 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2623 Commits where accidentially squashed together
2625 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2627 * gst/rtsp-server/rtsp-stream.c:
2628 * tests/check/gst/client.c:
2629 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2630 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2631 Commits where accidentially squashed together
2633 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2635 * gst/rtsp-server/rtsp-client.c:
2636 * gst/rtsp-server/rtsp-stream.c:
2637 * gst/rtsp-server/rtsp-stream.h:
2638 * tests/check/gst/client.c:
2639 * tests/check/gst/stream.c:
2640 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2641 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2642 Commits where accidentially squashed together
2644 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2646 * examples/test-auth-digest.c:
2647 examples: Fix indentation
2649 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2651 * gst/rtsp-server/rtsp-client.c:
2652 * gst/rtsp-server/rtsp-stream.c:
2653 * gst/rtsp-server/rtsp-stream.h:
2654 * tests/check/gst/client.c:
2655 * tests/check/gst/stream.c:
2656 stream: Choose the maximum ttl value provided by multicast clients
2657 The maximum ttl value provided so far by the multicast clients
2658 will be chosen and reported in the response to the current
2660 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2662 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2664 * gst/rtsp-server/rtsp-stream.c:
2665 * tests/check/gst/client.c:
2666 rtsp-stream: Don't require address pool in the transport specific case
2667 If "transport.client-settings" parameter is set to true, the client is
2668 allowed to specify destination, ports and ttl.
2669 There is no need for pre-configured address pool.
2670 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2672 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2674 * gst/rtsp-server/rtsp-client.c:
2675 * gst/rtsp-server/rtsp-media-factory.c:
2676 * gst/rtsp-server/rtsp-media-factory.h:
2677 * gst/rtsp-server/rtsp-media.c:
2678 * gst/rtsp-server/rtsp-media.h:
2679 * gst/rtsp-server/rtsp-stream.c:
2680 * gst/rtsp-server/rtsp-stream.h:
2681 * tests/check/gst/client.c:
2682 * tests/check/gst/mediafactory.c:
2683 Add new API for setting/getting maximum multicast ttl value
2684 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2686 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2688 * gst/rtsp-server/rtsp-stream.c:
2689 rtsp-stream: avoid duplicating the first multicast client
2690 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2691 clients were dynamically added and removed to the multicast
2692 udp sinks, as such we should no longer add a first client in
2693 set_multicast_socket_for_udpsink
2694 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2696 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2698 * gst/rtsp-server/Makefile.am:
2699 rtsp-server: Add gstreamer-base gir dir in autotools
2701 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2703 * gst/rtsp-server/rtsp-client.c:
2704 * gst/rtsp-server/rtsp-stream.c:
2705 rtsp-client: always allocate both IPV4 and IPV6 sockets
2706 multiudpsink does not support setting the socket* properties
2707 after it has started, which meant that rtsp-server could no
2708 longer serve on both IPV4 and IPV6 sockets since the patches
2709 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2711 When first connecting an IPV6 client then an IPV4 client,
2712 multiudpsink fell back to using the IPV6 socket.
2713 When first connecting an IPV4 client, then an IPV6 client,
2714 multiudpsink errored out, released the IPV4 socket, then
2715 crashed when trying to send a message on NULL nevertheless,
2716 that is however a separate issue.
2717 This could probably be fixed by handling the setting of
2718 sockets in multiudpsink after it has started, that will
2719 however be a much more significant effort.
2720 For now, this commit simply partially reverts the behaviour
2721 of rtsp-stream: it will continue to only create the udpsinks
2722 when needed, as was the case since the patches were merged,
2723 it will however when creating them, always allocate both
2724 sockets and set them on the sink before it starts, as was
2725 the case prior to the patches.
2726 Transport configuration will only error out if the allocation
2727 of UDP sockets fails for the actual client's family, this
2728 also downgrades the GST_ERRORs in alloc_ports_one_family
2729 to GST_WARNINGs, as failing to allocate is no longer
2731 https://bugzilla.gnome.org/show_bug.cgi?id=796875
2733 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2736 * meson_options.txt:
2737 meson: Convert common options to feature options
2738 These are necessary for gst-build to set options correctly. The
2739 remaining automagic option is cgroup support in examples.
2740 https://bugzilla.gnome.org/show_bug.cgi?id=795107
2742 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2744 * gst/rtsp-server/rtsp-stream.c:
2745 rtsp-stream: Slightly simplify locking
2747 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
2749 * gst/rtsp-server/rtsp-client.c:
2750 * gst/rtsp-server/rtsp-stream-transport.c:
2751 * gst/rtsp-server/rtsp-stream-transport.h:
2752 * gst/rtsp-server/rtsp-stream.c:
2753 Limit queued TCP data messages to one per stream
2754 Before, the watch backlog size in GstRTSPClient was changed
2755 dynamically between unlimited and a fixed size, trying to avoid both
2756 unlimited memory usage and deadlocks while waiting for place in the
2757 queue. (Some of the deadlocks were described in a long comment in
2759 In the previous commit, we changed to a fixed backlog size of 100.
2760 This is possible, because we now handle RTP/RTCP data messages differently
2761 from RTSP request/response messages.
2762 The data messages are messages tunneled over TCP. We allow at most one
2763 queued data message per stream in GstRTSPClient at a time, and
2764 successfully sent data messages are acked by sending a "message-sent"
2765 callback from the GstStreamTransport. Until that ack comes, the
2766 GstRTSPStream does not call pull_sample() on its appsink, and
2767 therefore the streaming thread in the pipeline will not be blocked
2768 inside GstRTSPClient, waiting for a place in the queue.
2769 pull_sample() is called when we have both an ack and a "new-sample"
2770 signal from the appsink. Then, we know there is a buffer to write.
2771 RTSP request/response messages are not acked in the same way as data
2772 messages. The rest of the 100 places in the queue are used for
2773 them. If the queue becomes full of request/response messages, we
2774 return an error and close the connection to the client.
2775 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2777 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
2779 * gst/rtsp-server/rtsp-client.c:
2780 rtsp-client: Use fixed backlog size
2781 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
2782 Preparation for the next commit, which changes to a different way of
2783 avoiding both deadlocks and unlimited memory usage with the watch
2786 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2788 * gst/rtsp-server/rtsp-media.c:
2789 rtsp-media: unref clock (if set) when finalizing
2790 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2792 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2794 * docs/libs/gst-rtsp-server-sections.txt:
2795 rtsp-media: add gst_rtsp_media_*_set_clock to docs
2796 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2798 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2800 * gst/rtsp-server/rtsp-media-factory.c:
2801 media-factory: unref old clock when setting new clock
2802 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2804 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
2806 * gst/rtsp-server/rtsp-media-factory.c:
2807 media-factory: unref clock in finalize
2808 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2810 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2812 * gst/rtsp-server/rtsp-onvif-media.c:
2813 rtsp-onvif-media: fix g-ir-scanner warnings
2815 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2818 .gitignore: add another example binary
2820 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
2822 * examples/meson.build:
2823 meson: add new test-appsrc2 example to meson build
2825 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2827 * examples/Makefile.am:
2828 examples: fix build of new test-appsrc2 example
2829 Need to link against libgstapp-1.0.
2831 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
2833 * examples/.gitignore:
2834 * examples/Makefile.am:
2835 * examples/test-appsrc2.c:
2836 examples: Add test-appsrc2
2837 Add an example of feeding both audio and video into an RTSP
2838 pipeline via appsrc.
2840 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
2842 * gst/rtsp-server/rtsp-client.c:
2843 client: Strip transport parts as whitespaces could be around commas
2844 https://bugzilla.gnome.org/show_bug.cgi?id=758428
2846 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
2848 * gst/rtsp-server/rtsp-stream.c:
2849 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
2850 Fix race when setting up source elements.
2851 Since we set the source element(s) to PLAYING state before hooking
2852 them up to the downstream funnel, it's possible for the source element
2853 to receive packets before we actually get to linking it to the funnel,
2854 in which case buffers would be pushed out on an unlinked pad, causing
2855 it to error out and stop receiving more data.
2856 We fix this by blocking the source's srcpad until we have linked it.
2857 https://bugzilla.gnome.org/show_bug.cgi?id=796160
2859 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
2861 * gst/rtsp-server/rtsp-stream.c:
2862 rtsp-stream: Fix mismatch between allowed and configured protocols
2863 https://bugzilla.gnome.org/show_bug.cgi?id=796679
2865 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
2867 * gst/rtsp-server/rtsp-stream.c:
2868 rtsp-stream: Emit a signal when the SRTP decoder is created
2869 https://bugzilla.gnome.org/show_bug.cgi?id=778080
2871 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
2873 * gst/rtsp-server/rtsp-stream.c:
2874 rtsp-stream: Don't require presence of sinks in _get_*_socket()
2875 Transport specific sink elements are added to the pipeline
2876 in PLAY request and sockets are already created in SETUP so
2877 it's actually wrong to require the presence of sinks in
2878 _get_*_socket() functions.
2879 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2881 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
2883 * gst/rtsp-server/rtsp-stream.c:
2884 rtsp-stream: Update transport for multicast clients as well
2885 If a multicast client requests different transport settings
2886 than the existing one make sure that this new transport
2887 configuruation is propagated to the multicast udp sink.
2888 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2890 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
2892 * gst/rtsp-server/rtsp-stream.c:
2893 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
2894 And not on unicast udp sinks
2895 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2897 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
2899 * gst/rtsp-server/rtsp-address-pool.c:
2900 * gst/rtsp-server/rtsp-auth.c:
2901 * gst/rtsp-server/rtsp-client.c:
2902 * gst/rtsp-server/rtsp-media-factory-uri.c:
2903 * gst/rtsp-server/rtsp-media-factory.c:
2904 * gst/rtsp-server/rtsp-media.c:
2905 * gst/rtsp-server/rtsp-mount-points.c:
2906 * gst/rtsp-server/rtsp-server.c:
2907 * gst/rtsp-server/rtsp-session-media.c:
2908 * gst/rtsp-server/rtsp-session-pool.c:
2909 * gst/rtsp-server/rtsp-session.c:
2910 * gst/rtsp-server/rtsp-stream-transport.c:
2911 * gst/rtsp-server/rtsp-stream.c:
2912 * gst/rtsp-server/rtsp-thread-pool.c:
2913 Update for g_type_class_add_private() deprecation in recent GLib
2915 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
2917 * gst/rtsp-server/rtsp-auth.c:
2918 * gst/rtsp-server/rtsp-media.c:
2919 * gst/rtsp-server/rtsp-sdp.c:
2920 * gst/rtsp-server/rtsp-stream.c:
2923 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
2925 * examples/Makefile.am:
2926 * examples/test-video-disconnect.c:
2927 examples: Add test-video-disconnect example
2928 Simple example which cuts off all clients 10 seconds
2929 after the first one connects.
2931 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2933 * docs/libs/gst-rtsp-server-sections.txt:
2934 * examples/test-auth-digest.c:
2935 * gst/rtsp-server/rtsp-auth.c:
2936 * gst/rtsp-server/rtsp-auth.h:
2937 rtsp-auth: Add support for parsing .htdigest files
2938 Passwords are usually not stored in clear text, but instead
2939 stored already hashed in a .htdigest file.
2940 Add support for parsing such files, add API to allow setting
2941 a custom realm in RTSPAuth, and update the digest example.
2942 https://bugzilla.gnome.org/show_bug.cgi?id=796637
2944 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
2946 * gst/rtsp-sink/gstrtspclientsink.c:
2947 * gst/rtsp-sink/gstrtspclientsink.h:
2948 rtspclientsink: fix waiting for multiple streams
2949 We were previously only ever waiting for a single stream to notify it's
2950 blocked status through GstRTSPStreamBlocking. Actually count streams to
2952 Fixes rtspclientsink sending SDP's without out some of the input
2954 https://bugzilla.gnome.org/show_bug.cgi?id=796624
2956 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2958 * docs/libs/gst-rtsp-server-sections.txt:
2959 docs: add missing auth methods
2961 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2963 * gst/rtsp-server/rtsp-stream.c:
2964 rtsp-stream: only create funnel if it didn't exist already.
2965 This precented using multiple protocols for the same stream.
2966 https://bugzilla.gnome.org/show_bug.cgi?id=796634
2968 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2970 * examples/meson.build:
2971 meson: build auth-digest example
2973 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
2975 * gst/rtsp-server/rtsp-client.c:
2976 * gst/rtsp-server/rtsp-media.c:
2977 * gst/rtsp-server/rtsp-sdp.c:
2978 * gst/rtsp-server/rtsp-session-media.c:
2979 * gst/rtsp-server/rtsp-stream-transport.c:
2980 Get payloader stats only for the sending streams
2981 Get/set payloader properties only for streams that actually
2982 contain a payloader element.
2983 https://bugzilla.gnome.org/show_bug.cgi?id=796523
2985 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
2987 * gst/rtsp-server/Makefile.am:
2988 Makefile: Don't hardcode libtool for g-i build
2989 Similar to the other commits in core/base/bad
2991 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
2993 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2994 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
2995 https://bugzilla.gnome.org/show_bug.cgi?id=796229
2997 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
2999 * gst/rtsp-sink/gstrtspclientsink.c:
3000 rtspclientsink: Don't deadlock in preroll on early close
3001 If the connection is closed very early, the flushing
3002 marker might not get set and rtspclientsink can get
3003 deadlocked waiting for preroll forever.
3004 https://bugzilla.gnome.org/show_bug.cgi?id=786961
3006 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3009 * meson_options.txt:
3010 meson: Update option names to omit disable_ and with- prefixes
3011 Also yield common options to the outer project (gst-build in our case)
3012 so that they don't have to be set manually.
3014 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3017 meson: use -Wl,-Bsymbolic-functions where supported
3018 Just like the autotools build.
3020 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3023 * tests/check/Makefile.am:
3024 configure: check for -good and -bad plugins only in uninstalled setup
3025 Avoids confusing configure messages looking or a -good .pc file
3027 Also use plugindir variables that common macros set while at it.
3028 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3030 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3032 * gst/rtsp-server/rtsp-client.c:
3033 rtsp-client: Fix session timeout
3034 When streaming data over TCP then is not the keep-alive
3035 functionality working.
3036 The reason is that the function do_send_data have changed
3037 to boolean but the code is still checking the received result
3038 from send_func with GST_RTSP_OK.
3039 The result is that a successful send_func will always lead to
3040 that do_send_data is returning false and the keep-alive will
3042 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3044 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3046 * docs/libs/gst-rtsp-server-sections.txt:
3047 * gst/rtsp-server/rtsp-media.c:
3048 * gst/rtsp-server/rtsp-sdp.c:
3049 * gst/rtsp-server/rtsp-stream.c:
3050 * gst/rtsp-server/rtsp-stream.h:
3051 * gst/rtsp-sink/gstrtspclientsink.c:
3052 * gst/rtsp-sink/gstrtspclientsink.h:
3053 Implement support for ULP Forward Error Correction
3054 In this initial commit, interface is only exposed for RECORD,
3055 further work will be needed in rtspsrc to support this for
3057 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3059 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3061 * gst/rtsp-server/rtsp-onvif-media.c:
3062 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3063 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3064 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3065 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3066 the opposite, just like the ONVIF standard.
3067 Let's follow those RFCs as we're doing RTSP here, and add a property at
3068 a later time if needed to switch to the SDP RFC behaviour.
3069 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3071 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3074 Automatic update of common submodule
3075 From 3fa2c9e to ed78bee
3077 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3079 * gst/rtsp-server/rtsp-client.c:
3080 * gst/rtsp-server/rtsp-media-factory.c:
3081 * gst/rtsp-server/rtsp-media.c:
3082 * gst/rtsp-server/rtsp-stream.c:
3083 * tests/check/gst/rtspclientsink.c:
3084 gst: Run everything through gst-indent again
3086 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3088 * gst/rtsp-server/rtsp-media.c:
3089 * tests/check/gst/media.c:
3090 rtsp-media: query the position on active streams if media is complete
3091 If the media is complete, i.e. one or more streams have been configured
3092 with sinks, then we want to query the position on those streams only.
3093 A query on an incomplete stream may return a position that originates from
3095 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3097 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3099 * gst/rtsp-sink/gstrtspclientsink.c:
3100 rtspclientsink: make sure not to use freed string
3101 Set transport string to NULL after freeing it, so that
3102 at worst we get a NULL pointer if constructing a new
3103 transport string fails (which shouldn't really fail here).
3104 Also check return value of that, just in case.
3107 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3109 * gst/rtsp-server/rtsp-client.c:
3110 rtsp-client: do not free string passed to take_header
3112 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3114 * gst/rtsp-server/rtsp-stream.c:
3115 rtsp-stream: do not take lock in request_aux_receiver
3116 Added it right before pushing the previous commit, it is
3117 incorrect and deadlocks because this function gets called
3118 from the join_bin thread, which already holds the lock,
3119 that's the reason why request_aux_sender didn't take the
3122 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3124 * docs/libs/gst-rtsp-server-sections.txt:
3125 * gst/rtsp-server/rtsp-media-factory.c:
3126 * gst/rtsp-server/rtsp-media-factory.h:
3127 * gst/rtsp-server/rtsp-media.c:
3128 * gst/rtsp-server/rtsp-media.h:
3129 * gst/rtsp-server/rtsp-stream.c:
3130 * gst/rtsp-server/rtsp-stream.h:
3131 rtsp-server: add API to enable retransmission requests
3132 "do-retransmission" was previously set when rtx-time != 0,
3133 which made no sense as do-retransmission is used to enable
3134 the sending of retransmission requests, where as rtx-time
3135 is used by the peer to enable storing of buffers in order
3136 to respond to retransmission requests.
3137 rtsp-media now also provides a callback for the
3138 request-aux-receiver signal.
3139 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3141 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3143 * gst/rtsp-sink/gstrtspclientsink.c:
3144 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3145 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3147 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3149 * gst/rtsp-sink/gstrtspclientsink.c:
3150 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3151 This in order to be able to decrypt the RTCP backchannel
3152 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3154 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3156 * gst/rtsp-server/rtsp-client.c:
3157 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3158 When sending back an encrypted RTCP back channel, it is useful
3159 for the client to know the encryption key.
3160 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3162 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3164 * gst/rtsp-server/rtsp-client.c:
3165 * gst/rtsp-server/rtsp-stream.c:
3166 * gst/rtsp-server/rtsp-stream.h:
3167 rtsp-stream: extract handle_keymgmt from rtsp-client
3168 rtspclientsink will also need to parse KeyMgmt headers
3169 sent by the server to decrypt the RTCP backchannel stream
3170 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3172 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3174 * gst/rtsp-sink/gstrtspclientsink.c:
3175 * tests/check/gst/rtspclientsink.c:
3176 rtspclientsink: Fix client ports for the RTCP backchannel
3177 This was broken since the work for delayed transport creation
3178 was merged: the creation of the transports string depends on
3179 calling stream_get_server_port, which only starts returning
3180 something meaningful after a call to stream_allocate_udp_sockets
3181 has been made, this function expects a transport that we parse
3182 from the transport string ...
3183 Significant refactoring is in order, but does not look entirely
3184 trivial, for now we put a band aid on and create a second transport
3185 string after the stream has been completed, to pass it in
3186 the request headers instead of the previous, incomplete one.
3187 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3189 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3191 * gst/rtsp-server/rtsp-client.c:
3192 rtsp-client:Error handling when equal http session cookie
3193 There are some clients that are sending same session cookie on random
3195 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3197 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3199 * gst/rtsp-server/rtsp-media-factory-uri.c:
3200 rtsp-media-factory-uri: Fix compilation with latest GLib
3201 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3202 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3203 data->factory = g_object_ref (factory);
3206 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3214 === release 1.14.0 ===
3216 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3222 * gst-rtsp-server.doap:
3226 === release 1.13.91 ===
3228 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3234 * gst-rtsp-server.doap:
3238 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3240 * gst/rtsp-server/Makefile.am:
3241 * gst/rtsp-server/meson.build:
3242 * gst/rtsp-server/rtsp-address-pool.h:
3243 * gst/rtsp-server/rtsp-auth.h:
3244 * gst/rtsp-server/rtsp-client.h:
3245 * gst/rtsp-server/rtsp-context.h:
3246 * gst/rtsp-server/rtsp-media-factory-uri.h:
3247 * gst/rtsp-server/rtsp-media-factory.h:
3248 * gst/rtsp-server/rtsp-media.h:
3249 * gst/rtsp-server/rtsp-mount-points.h:
3250 * gst/rtsp-server/rtsp-onvif-client.h:
3251 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3252 * gst/rtsp-server/rtsp-onvif-media.h:
3253 * gst/rtsp-server/rtsp-onvif-server.h:
3254 * gst/rtsp-server/rtsp-params.h:
3255 * gst/rtsp-server/rtsp-permissions.h:
3256 * gst/rtsp-server/rtsp-sdp.h:
3257 * gst/rtsp-server/rtsp-server-prelude.h:
3258 * gst/rtsp-server/rtsp-server.h:
3259 * gst/rtsp-server/rtsp-session-media.h:
3260 * gst/rtsp-server/rtsp-session-pool.h:
3261 * gst/rtsp-server/rtsp-session.h:
3262 * gst/rtsp-server/rtsp-stream-transport.h:
3263 * gst/rtsp-server/rtsp-stream.h:
3264 * gst/rtsp-server/rtsp-thread-pool.h:
3265 * gst/rtsp-server/rtsp-token.h:
3266 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3267 We need different export decorators for the different libs.
3268 For now no actual change though, just rename before the release,
3269 and add prelude headers to define the new decorator to GST_EXPORT.
3271 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3273 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3274 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3275 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3277 === release 1.13.90 ===
3279 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3285 * gst-rtsp-server.doap:
3289 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3291 * gst/rtsp-server/rtsp-media-factory.c:
3292 * gst/rtsp-server/rtsp-permissions.c:
3293 permissions: add Since tags and example for new API
3295 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3297 * docs/libs/gst-rtsp-server-sections.txt:
3298 * gst/rtsp-server/rtsp-media-factory.c:
3299 * gst/rtsp-server/rtsp-media-factory.h:
3300 * gst/rtsp-server/rtsp-permissions.c:
3301 * gst/rtsp-server/rtsp-permissions.h:
3302 * tests/check/gst/permissions.c:
3303 permissions: more bindings-friendly API
3304 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3306 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3309 meson: enable more warnings
3311 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3313 * gst/rtsp-server/rtsp-client.c:
3314 rtsp-client: Place netaddress meta on packets received via TCP
3315 This allows us to later map signals from rtpbin/rtpsource back to the
3316 corresponding stream transport, and allows to do keep-alive based on
3317 RTCP packets in case of TCP media transport.
3318 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3320 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3322 * gst/rtsp-sink/gstrtspclientsink.c:
3323 rtspclientsink: if OPEN failed, unqueue next command
3324 As READY_TO_PAUSED can no longer return async, the RECORD
3325 command will be queued before the OPEN command fails
3326 (for example in case the server could not be connected),
3327 and record then waits for ever.
3328 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3330 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3332 * gst/rtsp-sink/gstrtspclientsink.c:
3333 rtspclientsink: fix retrieval of custom payloader caps
3334 If a bin is passed as the custom payloader, the caps of
3335 its factory will be empty, the correct way to obtain the caps
3336 is to query its sinkpad.
3338 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3340 * gst/rtsp-sink/gstrtspclientsink.c:
3341 rtspclientsink: fix extra unref of custom payloader
3343 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3345 * gst/rtsp-sink/gstrtspclientsink.c:
3346 rspclientsink: fix recent code indentation
3348 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3350 * gst/rtsp-sink/gstrtspclientsink.c:
3351 rtspclientsink: add missing get_type prototype
3353 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3355 * gst/rtsp-sink/gstrtspclientsink.c:
3356 rtspclientsink: allow setting payloader as pad property
3357 This was a FIXME item, and can be quite useful, also
3358 allowing to specify payloader properties from the command
3359 line, which is always nice.
3360 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3362 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3364 * gst/rtsp-server/rtsp-media.c:
3365 rtsp-media: Replace g_print() log line
3366 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3368 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3370 * gst/rtsp-server/rtsp-media.c:
3371 * tests/check/gst/rtspclientsink.c:
3372 rtsp-media: fix RECORD getting stuck
3373 The test_record case was working because async=false had
3374 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3375 but that was incorrect, as it should not be needed.
3376 Removing async=false made the test fail as expected, this is
3377 fixed by not trying to preroll when preparing the media for
3378 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3379 and our peer will not start sending media until it has received
3380 a response to that request, and sent and received a response
3381 to RECORD as well, thus obviously preventing preroll.
3382 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3384 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3386 * gst/rtsp-server/rtsp-auth.c:
3387 rtsp-auth: fix set_tls_authentication_mode annotation
3389 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3391 * gst/rtsp-server/rtsp-onvif-media.c:
3392 rtp-server: remove redefined variable
3393 res is a boolean variable which is defined in the function scope and
3394 redefined, with no reason, in the loop scope. This patch removes the
3396 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3398 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3400 * gst/rtsp-server/rtsp-media.c:
3401 * gst/rtsp-server/rtsp-stream.c:
3402 * gst/rtsp-server/rtsp-stream.h:
3403 stream: Add functions for checking if stream is receiver or sender
3404 ...and replace all checks for RECORD in GstRTSPMedia which are really
3405 for "sender-only". This way the code becomes more generic and introducing
3406 support for onvif-backchannel later on will require no changes in
3409 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3411 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3412 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3413 onvif: Make requires_backchannel() public
3414 ...in order to let subclasses building the onvif part of the pipeline
3415 check whether backchannel shall be included or not.
3417 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3419 * gst/rtsp-server/rtsp-onvif-media.c:
3420 rtsp-server: Switch around sendonly/recvonly attributes
3421 They are wrong in the ONVIF streaming spec. The backchannel should be
3422 recvonly and the normal media should be sendonly: direction is always
3423 from the point of view of the SDP offerer (the server) according to
3426 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3428 * docs/libs/gst-rtsp-server-docs.sgml:
3429 * docs/libs/gst-rtsp-server-sections.txt:
3430 * examples/.gitignore:
3431 * examples/Makefile.am:
3432 * examples/test-onvif-backchannel.c:
3433 * gst/rtsp-server/Makefile.am:
3434 * gst/rtsp-server/rtsp-media.h:
3435 * gst/rtsp-server/rtsp-onvif-client.c:
3436 * gst/rtsp-server/rtsp-onvif-client.h:
3437 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3438 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3439 * gst/rtsp-server/rtsp-onvif-media.c:
3440 * gst/rtsp-server/rtsp-onvif-media.h:
3441 * gst/rtsp-server/rtsp-onvif-server.c:
3442 * gst/rtsp-server/rtsp-onvif-server.h:
3443 * gst/rtsp-server/rtsp-sdp.c:
3444 * gst/rtsp-server/rtsp-sdp.h:
3445 rtsp: Add support for ONVIF backchannel
3446 This adds a new RTSP server, client, media-factory and media subclass
3447 for handling the specifics of the backchannel. Ideally this later can be
3448 extended with other ONVIF specific features.
3450 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3452 * gst/rtsp-server/rtsp-media.c:
3453 rtsp-media: Add support for sending+receiving medias
3454 We need to add an appsrc/appsink in that case because otherwise the
3455 media bin will be a sink and a source for rtpbin, causing a pipeline
3457 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3459 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3465 === release 1.13.1 ===
3467 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3471 * gst-rtsp-server.doap:
3475 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3477 * gst/rtsp-server/rtsp-session-pool.c:
3478 session-pool: remove nullable return annotation
3479 create_watch can only return NULL from the API guards, no
3482 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3484 * gst/rtsp-server/rtsp-media-factory.c:
3485 * gst/rtsp-server/rtsp-media.c:
3486 set_clock functions: Add nullable annotations
3488 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3490 * gst/rtsp-server/rtsp-auth.c:
3491 * gst/rtsp-server/rtsp-client.c:
3492 * gst/rtsp-server/rtsp-media-factory.c:
3493 * gst/rtsp-server/rtsp-media.c:
3494 * gst/rtsp-server/rtsp-mount-points.c:
3495 * gst/rtsp-server/rtsp-server.c:
3496 * gst/rtsp-server/rtsp-session-media.c:
3497 * gst/rtsp-server/rtsp-session-pool.c:
3498 * gst/rtsp-server/rtsp-session.c:
3499 * gst/rtsp-server/rtsp-stream-transport.c:
3500 * gst/rtsp-server/rtsp-stream.c:
3501 * gst/rtsp-server/rtsp-thread-pool.c:
3502 All around: add annotations and API guards
3504 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3506 * tests/test-cleanup.c:
3507 test-cleanup: bind any port
3508 The meson test suite runs tests in parallel, trying to bind
3509 a single port made the test fail.
3511 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3514 meson: make version numbers ints and fix int/string comparison
3515 WARNING: Trying to compare values of different types (str, int).
3516 The result of this is undefined and will become a hard error
3517 in a future Meson release.
3519 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3521 * gst/rtsp-server/rtsp-context.c:
3522 gst_rtsp_context_get_current: add (skip) annotation
3523 The return value type is defined with G_DEFINE_POINTER_TYPE,
3524 and gi emits the following warning:
3525 Invalid non-constant return of bare structure or union; register as
3526 boxed type or (skip)
3528 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3530 * gst/rtsp-server/rtsp-client.c:
3531 rtsp-client: add type annotations
3532 gi doesn't seem to be able to figure out the type of the
3533 signal parameters when defined with G_DEFINE_POINTER_TYPE
3535 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3538 autotools: use -fno-strict-aliasing where supported
3539 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3541 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3544 meson: use -fno-strict-aliasing where supported
3545 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3547 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3549 * gst/rtsp-server/rtsp-mount-points.c:
3550 mount-points: bail out of loop again when matching mount points
3551 Previous patch led to us iterating the entire sequence. Bail out
3552 of the loop again if we have a match but are moving away from it.
3553 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3555 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3557 * tests/check/gst/mountpoints.c:
3558 tests: mountpoints: add more checks for mount point path matching
3559 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3561 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3563 * gst/rtsp-server/rtsp-mount-points.c:
3564 mount-points: fix matching of paths where there's also an entry with a common prefix
3565 e.g. with the following mount points
3569 _match() would not match /raw/video and /raw/snapshot correctly.
3570 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3572 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3574 * docs/libs/gst-rtsp-server-sections.txt:
3575 * gst/rtsp-server/rtsp-permissions.c:
3576 * gst/rtsp-server/rtsp-permissions.h:
3577 * tests/check/gst/permissions.c:
3578 permissions: add some new API to make this usable from bindings
3579 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3581 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3583 * gst/rtsp-server/rtsp-token.c:
3584 rtsp-token: annotate constructors for bindings
3585 This maps _new_empty() to _new(), which also makes RTSPToken()
3586 work properly now. Since this API wasn't usable from bindings
3587 before, this should hopefully be fine.
3588 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3590 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3592 * docs/libs/gst-rtsp-server-sections.txt:
3593 * gst/rtsp-server/rtsp-token.c:
3594 * gst/rtsp-server/rtsp-token.h:
3595 * tests/check/gst/token.c:
3596 rtsp-token: add some API to set fields from bindings
3597 The existing functions are all vararg-based and as such
3598 not usable from bindings.
3599 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3601 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3603 * tests/check/gst/rtspclientsink.c:
3604 * tests/check/gst/rtspserver.c:
3605 * tests/check/gst/sessionpool.c:
3606 * tests/check/gst/stream.c:
3607 tests: fix indentation
3610 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3612 * tests/check/gst/rtspserver.c:
3613 tests: rtspserver: fix another ref leak
3614 Even if this didn't show up in valgrind.
3616 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3618 * tests/check/gst/rtspclientsink.c:
3619 tests: rtspclientsink: fix leak
3621 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3623 * tests/check/gst/rtspserver.c:
3624 test: rtspserver: plug memory leak in test_no_session_timeout
3625 In test_no_session_timeout, unref the rtsp session object when the
3627 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3629 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3631 * gst/rtsp-sink/gstrtspclientsink.c:
3632 rtpsclientsink: Initialize and clear newly added mutex and cond
3633 While it *did* work, glib would automatically create new mutex and cond
3634 ... which never got freed
3636 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3638 * gst/rtsp-server/rtsp-stream.c:
3639 rtsp-stream: Set multicast TTL on the multicast sockets
3640 And not if we do unicast UDP.
3641 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3643 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3645 * gst/rtsp-server/rtsp-stream.c:
3646 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3647 In the multicast case (as in test-multicast, not test-multicast2), the
3648 address could be allocated/reserved (and thus set) already without
3649 allocating the actual socket. We need to allocate the socket here still
3650 instead of just claiming that it was already allocated.
3651 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3653 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3655 * gst/rtsp-sink/gstrtspclientsink.c:
3656 * gst/rtsp-sink/gstrtspclientsink.h:
3657 rtspclientsink: Use the new rtsp-stream API
3658 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3660 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3662 * gst/rtsp-sink/gstrtspclientsink.c:
3663 * gst/rtsp-sink/gstrtspclientsink.h:
3664 rtspclientsink: Wait until OPEN has been scheduled
3665 Make sure that the sink thread has started opening connection
3666 to the server before continuing.
3667 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3669 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3672 Automatic update of common submodule
3673 From e8c7a71 to 3fa2c9e
3675 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3677 * gst/rtsp-server/rtsp-media.c:
3678 * gst/rtsp-server/rtsp-session-media.c:
3679 * gst/rtsp-server/rtsp-stream.c:
3680 rtsp-server: Minor doc fixes
3683 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3686 * tests/Makefile.am:
3687 tests: disable all tests when --disable-tests is used
3688 Move conditional subdir include into top level.
3689 Based on patch by: Joel Holdsworth
3690 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3692 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3695 * meson_options.txt:
3696 * tests/meson.build:
3697 meson: build more tests and add options to disable tests and examples
3699 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3701 * gst/rtsp-server/rtsp-session.c:
3702 Fix build when -Werror=deprecated-declarations is on
3703 As gst_rtsp_session_next_timeout is deprecated.
3705 ../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]
3706 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3708 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3709 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3710 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3713 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3716 Automatic update of common submodule
3717 From 3f4aa96 to e8c7a71
3719 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3721 * tests/check/gst/media.c:
3722 check/media: Add seekability test case: not all streams are active
3723 Media contains two streams but only one is complete and prepared
3725 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3727 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3729 * gst/rtsp-server/rtsp-stream.c:
3730 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3731 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3733 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3735 * gst/rtsp-server/rtsp-media.c:
3736 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
3737 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3739 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
3742 meson: remove vs_module_defs_dir variable which is no longer needed
3744 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
3746 * gst/rtsp-server/rtsp-session.h:
3749 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
3752 * gst/rtsp-server/meson.build:
3754 * win32/common/libgstrtspserver.def:
3755 win32: remove .def file with exports
3756 They're no longer needed, symbol exporting is now explicit
3757 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
3759 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3762 autotools: stop controlling symbol visibility with -export-symbols-regex
3763 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
3764 This should result in consistent behaviour for the autotools and
3767 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
3769 * gst/rtsp-server/rtsp-media.h:
3770 * gst/rtsp-server/rtsp-server.h:
3771 * gst/rtsp-server/rtsp-session.c:
3772 * gst/rtsp-server/rtsp-session.h:
3773 rtsp-server: add missing GST_EXPORT and export deprecated funcs
3775 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
3777 * tests/check/gst/media.c:
3778 check: Add seekability testing on medias
3779 Make sure that once GstRTSPMedia are prepared they returned
3780 the expected seekability results
3781 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3783 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
3785 * docs/libs/gst-rtsp-server-sections.txt:
3786 * gst/rtsp-server/rtsp-media.c:
3787 * gst/rtsp-server/rtsp-stream.c:
3788 * gst/rtsp-server/rtsp-stream.h:
3789 * win32/common/libgstrtspserver.def:
3790 rtsp-media: Enable seeking query before pipeline is complete
3791 SDP are now provided *before* the pipeline is fully complete. In order
3792 to know whether a media is seekable or not therefore requires asking
3793 the invididual streams.
3794 API: gst_rtsp_stream_seekable
3795 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3797 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
3799 * gst/rtsp-server/rtsp-media.c:
3800 rtsp-media: Fix handling in default_unsuspend()
3801 Handle the case when streams are not blocked and media
3802 is suspended from PAUSED.
3803 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
3804 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3806 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
3808 * tests/check/gst/media.c:
3809 check/media: Fix thread pool leak.
3810 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
3811 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3813 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
3815 * gst/rtsp-server/rtsp-media.c:
3816 rtsp-media: Removed fakesink elements
3817 There is not need of adding fakesink elements to the media
3818 pipeline in the dynamic-payloader case.
3819 The media pipeline itself is dynamically updated with
3820 the receiver and sender parts that are based on the client
3821 transport information known after SETUP has been received.
3822 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
3823 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3825 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
3827 * gst/rtsp-server/rtsp-media.c:
3828 rtsp-media: Corrected ASYNC_DONE handling
3829 Media is complete when all the transport based parts are
3830 added to the media pipeline. At this point ASYNC_DONE is
3831 posted by the media pipeline and media is ready to enter
3833 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
3834 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3836 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
3838 * tests/check/gst/media.c:
3839 check/media: Check that prepared media can provide a SDP
3840 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
3842 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
3844 * gst/rtsp-server/rtsp-client.c:
3845 rtsp-client: Don't leak addr
3848 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
3850 * gst/rtsp-server/rtsp-client.c:
3851 * gst/rtsp-server/rtsp-session-media.c:
3852 * gst/rtsp-server/rtsp-stream.c:
3855 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
3857 * gst/rtsp-server/rtsp-media.c:
3858 rtsp-media: Don't unblock with remaining dynamic payloaders
3859 If we still have some dynamic paylaoders which haven't posted
3860 no-more-pads yet, don't go to PREPARED if one of the streams
3862 The risk was that we would end up not exposing/using all specified
3864 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
3865 then it will take a bit more time to start. But only if those 3
3866 conditions are present.
3867 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3869 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
3871 * gst/rtsp-server/rtsp-media.c:
3874 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
3876 * gst/rtsp-server/rtsp-media.c:
3877 rtsp-media: Don't set float on a gint64 variable
3878 Just use 0. Fixes 'undefined' behaviour from clang
3880 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
3882 * gst/rtsp-server/rtsp-media.c:
3883 rtsp-media: Fix previous commit
3884 We only want to count dynamic payloaders
3886 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
3888 * gst/rtsp-server/rtsp-media.c:
3889 * tests/check/gst/media.c:
3890 rtsp-media: Handle multiple dynamic elements
3891 If we have more than one dynamic payloader in the pipeline, we need
3892 to wait until the *last* one emits 'no-more-pads' before switching
3894 Failure to do so would result in a race where some of the streams
3895 wouldn't properly be prepared
3896 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3898 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
3900 * win32/common/libgstrtspserver.def:
3901 win32: Fix exported symbols list
3903 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3905 * gst/rtsp-server/rtsp-stream.c:
3906 rtsp-stream: Only update the RTP udpsink if it actually exists
3907 For send-only streams it does not exist, but the RTCP udpsink might.
3909 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
3911 * win32/common/libgstrtspserver.def:
3912 win32: Update exports
3914 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
3916 * gst/rtsp-server/rtsp-media.c:
3917 * gst/rtsp-server/rtsp-stream.c:
3918 * gst/rtsp-server/rtsp-stream.h:
3919 rtsp-media: seek on media pipelines that are complete
3920 Make sure that a seek is performed on pipelines that
3921 contain at least one sink element.
3922 Change-Id: Icf398e10add3191d104b1289de612412da326819
3923 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3925 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
3927 * gst/rtsp-server/rtsp-client.c:
3928 * gst/rtsp-server/rtsp-media.c:
3929 * gst/rtsp-server/rtsp-media.h:
3930 * gst/rtsp-server/rtsp-stream.c:
3931 * gst/rtsp-server/rtsp-stream.h:
3932 * tests/check/gst/client.c:
3933 * tests/check/gst/media.c:
3934 * tests/check/gst/rtspserver.c:
3935 * tests/check/gst/stream.c:
3936 Dynamically reconfigure pipeline in PLAY based on transports
3937 The initial pipeline does not contain specific transport
3938 elements. The receiver and the sender parts are added
3940 If the media is shared, the streams are dynamically
3941 reconfigured after each PLAY.
3942 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3944 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
3946 * gst/rtsp-server/rtsp-stream.c:
3947 rtsp-stream: obtain stream position from pad
3948 If no sinks have been added yet, obtain the current and
3949 the stop position of the stream from the send_src pad.
3950 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
3951 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3953 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
3955 * gst/rtsp-server/rtsp-session-media.c:
3956 * gst/rtsp-server/rtsp-session-media.h:
3957 rtsp-session-media: add function to get a list of transports
3958 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
3959 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3961 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
3963 * gst/rtsp-server/rtsp-stream.c:
3964 * gst/rtsp-server/rtsp-stream.h:
3965 rtsp-stream: add functions to get rtp and rtcp multicast sockets
3966 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
3967 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3969 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
3971 * gst/rtsp-server/rtsp-stream.c:
3972 stream: set async=sync=false only for RTCP appsink
3973 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
3974 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3976 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
3978 * gst/rtsp-server/rtsp-media.c:
3979 rtsp-media: return minimum value in query position case
3980 The minimum position should be returned as we are interested
3981 in the whole interval.
3982 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
3983 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3985 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
3987 * gst/rtsp-server/rtsp-session.c:
3988 * tests/check/gst/rtspserver.c:
3989 rtsp-session: Handle the case when timeout=0
3990 According to the documentation, a timeout of value 0 means
3991 that the session never timeouts. This adds handling of that.
3992 If timeout=0 we just return with a -1 from
3993 gst_rtsp_session_next_timeout_usec ().
3994 https://bugzilla.gnome.org/show_bug.cgi?id=785058
3996 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3998 * gst/rtsp-sink/gstrtspclientsink.c:
3999 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
4000 https://bugzilla.gnome.org/show_bug.cgi?id=785024
4002 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
4004 * docs/libs/gst-rtsp-server-sections.txt:
4005 * gst/rtsp-server/rtsp-media-factory.c:
4006 docs: add media factory transport mode accessors
4007 and fix the documentation for the return value of the getter
4009 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
4011 * gst/rtsp-server/rtsp-client.c:
4012 rtsp-client: unref 'pipelined_requests' in finalize
4013 The hash table priv->pipelined_requests is not unref:ed in the
4014 finalize funktion. Make sure it is.
4015 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4017 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4019 * gst/rtsp-server/rtsp-media.c:
4020 rtsp-media: Initialize scalar variable
4023 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4025 * win32/common/libgstrtspserver.def:
4026 win32: Update export file
4028 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4030 * gst/rtsp-server/rtsp-client.c:
4031 * gst/rtsp-server/rtsp-media.c:
4032 * gst/rtsp-server/rtsp-media.h:
4033 Start support for RTSP 2.0
4034 This adds basic support for new 2.0 features, though the protocol is
4035 subposdely backward incompatible, most semantics are the sames.
4038 * version negotiation
4039 * pipelined requests support
4040 * Media-Properties support
4041 * Accept-Ranges support
4043 * gst_rtsp_media_seekable
4044 The RTSP methods that have been removed when using 2.0 now return
4046 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4048 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4050 * gst/rtsp-server/rtsp-stream.c:
4051 stream: Use stream duration as stream-stop if segment was not configured with a stop
4052 Allowing client to know stream duration when no seeking happened.
4053 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4055 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4057 * gst/rtsp-server/rtsp-media-factory.c:
4058 rtsp-media-factory: Don't cache any media if NULL was returned as key
4059 The docs already mentioned this, but we actually stored it in the hash
4060 table with key==NULL and leaked its reference forever.
4062 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4064 * gst/rtsp-sink/gstrtspclientsink.c:
4065 * gst/rtsp-sink/gstrtspclientsink.h:
4066 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4067 This is a simple port of:
4068 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4069 * c438545dc9e2f14f657bc0ef261fff726449867b
4070 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4071 in gst-plugins-good.
4073 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4075 * gst/rtsp-server/rtsp-sdp.c:
4076 sdp: fix Memory leak in error case
4077 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4079 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4081 * pkgconfig/meson.build:
4082 meson: don't install -uninstalled.pc file
4083 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4085 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4088 Automatic update of common submodule
4089 From 48a5d85 to 3f4aa96
4091 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4093 * gst/rtsp-server/rtsp-client.c:
4094 rtsp-client: Fix typo in debug message
4096 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4099 meson: hide symbols by default unless explicitly exported
4101 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4103 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4104 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4105 Fixes meson warning about undefined @srcdir@.
4107 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4109 * tests/meson.build:
4110 meson: skip tests on windows for now
4111 As we do in the other modules. As libgstcheck is currently not
4112 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4113 the subproject 'gstreamer' does not exist"" Meson error.
4115 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4117 * gst/rtsp-server/rtsp-stream.c:
4118 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4119 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4120 multiudpsink's last-sample always comes from the payloader. Which
4121 is wrong if auxiliary streams are multiplexed in the same stream.
4122 So check the buffer's ssrc against the caps'ssrc before to use its
4123 seqnum. If not the same ssrc just use the payloader as done prior
4124 the commit above or when there is no last-sample yet.
4125 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4127 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4130 meson: Allow using glib as a subproject
4132 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4135 meson: fix with-package-name option
4136 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4138 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4141 Distribute meson_options.txt
4143 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4146 And config.h.meson is no longer dist either
4148 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4152 meson: config.h.meson is no longer needed
4154 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4156 * tests/check/meson.build:
4157 * tests/meson.build:
4158 meson: Fix building tests and activate them again
4160 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4162 * tests/check/meson.build:
4163 meson: Do not use path separator in test names
4164 Avoiding warnings like:
4165 WARNING: Target "elements/audioamplify" has a path separator in its name.
4167 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4170 * meson_options.txt:
4171 meson: add options to set package name and origin
4172 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4174 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4176 * gst/rtsp-server/rtsp-address-pool.h:
4177 * gst/rtsp-server/rtsp-auth.h:
4178 * gst/rtsp-server/rtsp-client.h:
4179 * gst/rtsp-server/rtsp-context.h:
4180 * gst/rtsp-server/rtsp-media-factory-uri.h:
4181 * gst/rtsp-server/rtsp-media-factory.h:
4182 * gst/rtsp-server/rtsp-media.h:
4183 * gst/rtsp-server/rtsp-mount-points.h:
4184 * gst/rtsp-server/rtsp-params.h:
4185 * gst/rtsp-server/rtsp-permissions.h:
4186 * gst/rtsp-server/rtsp-sdp.h:
4187 * gst/rtsp-server/rtsp-server.h:
4188 * gst/rtsp-server/rtsp-session-media.h:
4189 * gst/rtsp-server/rtsp-session-pool.h:
4190 * gst/rtsp-server/rtsp-session.h:
4191 * gst/rtsp-server/rtsp-stream-transport.h:
4192 * gst/rtsp-server/rtsp-stream.h:
4193 * gst/rtsp-server/rtsp-thread-pool.h:
4194 * gst/rtsp-server/rtsp-token.h:
4195 Mark symbols explicitly for export with GST_EXPORT
4197 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4200 * gst/rtsp-sink/Makefile.am:
4201 Remove plugin specific static build option
4202 Static and dynamic plugins now have the same interface. The standard
4203 --enable-static/--enable-shared toggle are sufficient.
4205 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4211 === release 1.12.0 ===
4213 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4219 * gst-rtsp-server.doap:
4223 === release 1.11.91 ===
4225 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4231 * gst-rtsp-server.doap:
4235 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4238 Automatic update of common submodule
4239 From 60aeef6 to 48a5d85
4241 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4243 * gst/rtsp-server/rtsp-media-factory.c:
4244 * gst/rtsp-server/rtsp-media.c:
4245 * gst/rtsp-server/rtsp-session.c:
4246 * gst/rtsp-server/rtsp-stream.c:
4247 gi: Fix some annotations and docstrings
4249 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4251 * gst/rtsp-server/meson.build:
4253 * meson_options.txt:
4256 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4260 Automatic update of common submodule
4261 From 39ac2f5 to 60aeef6
4263 === release 1.11.90 ===
4265 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4271 * gst-rtsp-server.doap:
4275 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4277 * examples/test-launch.c:
4278 examples: make test-launch pipeline shared by default as well
4280 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4282 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4283 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4284 Just the build dir is not going to work for srcdir!=builddir.
4286 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4289 meson: Update version
4291 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4296 === release 1.11.2 ===
4298 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4304 * gst-rtsp-server.doap:
4307 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4310 meson: dist meson build files
4311 Ship meson build files in tarballs, so people who use tarballs
4312 in their builds can start playing with meson already.
4314 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4316 * examples/test-record.c:
4317 examples/test-record: Add extra line to initial printout
4318 Add an example line of how to deliver a stream to the
4321 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4323 * gst/rtsp-server/rtsp-client.c:
4324 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4325 If there is no Content-Length header, no body would be allocated and the
4326 '\0' would also not be appended to the body.
4328 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4330 * gst/rtsp-server/rtsp-client.c:
4331 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4332 While they logically have 0 bytes length, GstRTSPConnection is appending
4333 a '\0' to everything making the size be 1 instead.
4335 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4340 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4342 * gst/rtsp-server/rtsp-session.c:
4343 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4344 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4347 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4352 === release 1.11.1 ===
4354 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4360 * gst-rtsp-server.doap:
4361 * win32/common/libgstrtspserver.def:
4364 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4366 * gst/rtsp-server/rtsp-stream.c:
4367 rtsp-stream: corrected if-statement in _get_server_port()
4368 This bug was accidentally introduced while fixing a segfault
4369 in _get_server_port() function.
4370 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4372 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4374 * gst/rtsp-server/rtsp-stream.c:
4375 * tests/check/gst/stream.c:
4376 rtsp-stream: fixed segmenation fault in _get_server_port()
4377 Calling function gst_rtsp_stream_get_server_port() results in
4378 segmenation fault in the RTP/RTSP/TCP case.
4379 Port that the server will use to receive RTCP makes only
4380 sense in the UDP case, however the function should handle
4381 the TCP case in a nicer way.
4382 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4384 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4386 * gst/rtsp-server/rtsp-media-factory.c:
4387 dosc: Fix a little typo
4388 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4390 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4392 * pkgconfig/Makefile.am:
4393 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4394 * pkgconfig/meson.build:
4395 meson: generate pkg-config -uninstalled pc files
4396 Generating those files is useful for users building the GStreamer stack
4397 using meson and having to link it to another project which is still
4398 using the autotools.
4399 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4401 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4403 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4404 pkgconfig: fix -uninstalled pc file
4405 pcfiledir was never defined so the paths were wrong.
4406 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4408 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4410 * gst/rtsp-server/rtsp-stream.c:
4411 * tests/check/gst/rtspserver.c:
4412 rtsp-stream: Fixed TCP transport case
4413 Make sure that the appsink element is actually added to
4414 the bin before trying to link it with the elements in it.
4415 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4417 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4423 Remove generated .spec file
4424 Likely extremely bitrotten, and we should not ship this anyway.
4426 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4429 Automatic update of common submodule
4430 From f980fd9 to 39ac2f5
4432 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4434 * gst/rtsp-server/rtsp-media.c:
4435 media: Fix pt map caps
4436 Since decryption is handled within rtpbin, all outcoming stream
4437 caps will be application/x-rtp (i.e. regular rtp)
4438 Fixes RECORD with SRTP streams
4440 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4442 * gst/rtsp-server/rtsp-media-factory.c:
4443 media-factory: Create media objects with the proper transport mode
4444 The function called immediately afterwards (collect_streams()) will
4445 need it to work properly
4447 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4449 * gst/rtsp-server/rtsp-auth.c:
4450 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4452 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4454 * gst/rtsp-server/rtsp-media-factory.c:
4455 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4456 We're going to put a pipeline into a pipeline otherwise, which is not
4459 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4461 * gst/rtsp-server/rtsp-media.c:
4462 media: Fix race condition around finish_unprepare() if called multiple time
4463 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4465 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4467 * gst/rtsp-sink/gstrtspclientsink.c:
4468 rtspclientsink: Don't leave stale pointer after unref
4469 Fix a warning on shutdown - don't keep a pointer to an
4470 alread-unreffed object.
4472 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4475 common: use https protocol for common submodule
4476 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4478 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4480 * gst/rtsp-server/rtsp-stream.c:
4481 stream: block the output of rtpbin instead of the source pipeline
4482 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4483 detection of the srtp rollover counter to add to the SDP.
4484 Unfortunately, it was incomplete for live pipelines where the logic
4485 blocks the source bin before creating the SDP and thus would never have
4486 the necessary informaiton to create a correct SDP with srtp encryption.
4487 Move the pad blocks to rtpbin's output pads instead so that the
4488 necessary information can be created before we need the information for
4490 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4492 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4494 * gst/rtsp-server/rtsp-client.c:
4495 rtsp-client: add IDLE timeout, before session exists
4496 The RTSP server will not timeout an idle RTSP connection
4497 (note this is different from doing timeout on a RTSP
4499 At least for Apache this is a problem when running RTSP over
4500 HTTPS since it uses one of the threads (there is a rather
4501 limited number) that are available for handling requests.
4502 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4504 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4509 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4511 * gst/rtsp-server/rtsp-stream.c:
4512 rtsp-stream: Set close-socket FALSE on UDP src:es
4513 With this RTSP server can use the sockets independent on the udpsrc
4515 When the udp src is finalized it will unref socket and when g_socket
4516 is finalized the socket will be closed.
4517 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4519 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4521 * gst/rtsp-sink/gstrtspclientsink.c:
4522 rtspclientsink: Move to new helper function to parse authentication responses
4523 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4525 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4527 * examples/Makefile.am:
4528 * examples/test-auth-digest.c:
4529 * gst/rtsp-server/rtsp-auth.c:
4530 * gst/rtsp-server/rtsp-auth.h:
4531 * win32/common/libgstrtspserver.def:
4532 rtsp-auth: Add support for Digest authentication
4533 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4535 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4538 * gst/rtsp-server/meson.build:
4540 * tests/check/meson.build:
4542 * win32/common/libgstrtspserver.def:
4543 Enable building with MSVC
4544 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4546 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4549 meson: gstreamer gst_check_dep does not exist on windows
4551 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4553 * gst/rtsp-server/rtsp-client.c:
4554 client: update do_send_message to match type GstRTSPClientSendFunc
4555 This type mismatch fails building with MSVC
4556 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4558 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4560 * gst/rtsp-server/rtsp-sdp.c:
4561 rtsp-sdp: Fix indentation
4563 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4565 * gst/rtsp-server/rtsp-media.c:
4566 rtsp-media: Only signal "new-state" if the state has actually changed
4567 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4569 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4571 * gst/rtsp-server/rtsp-client.c:
4572 * gst/rtsp-server/rtsp-client.h:
4573 client: emit signal in the beginning of each rtsp request
4574 These signals let the application validate the requests, configure the
4575 media/stream in a certain way and also generate error status code in
4576 case of error or bad request.
4577 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4579 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4582 meson: update version
4584 === release 1.11.0 ===
4586 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4591 === release 1.10.0 ===
4593 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4599 * gst-rtsp-server.doap:
4602 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4604 * tests/check/gst/rtspserver.c:
4605 * tests/check/gst/stream.c:
4606 tests: try to avoid using the same ports in different tests
4607 Causes problems with client multicast tests otherwise if
4608 tests are run in parallel.
4609 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4611 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4613 * tests/check/gst/client.c:
4614 tests: client: use fail_unless_equals_foo() for better failure reporting
4616 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4618 * gst/rtsp-server/rtsp-client.c:
4619 rtsp-client: Session filter in unwatch session
4620 Call session filter with filter_session_media as paramer in
4621 client_unwatch_session if using drop_backlog = FALSE.
4622 In client_unwatch_session its allowed to grow the watchs backlog.
4623 If using drop_backlog = FALSE and the backlog is full it will cause
4624 a deadlock when setting session media state to NULL
4625 if the backlog is not allowed to grow.
4626 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4628 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4631 meson: add fallbacks for gst modules
4634 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4636 * gst/rtsp-server/rtsp-client.c:
4637 rtsp-client: Fix factory leaking in find_media() in error cases
4638 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4640 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4642 * gst/rtsp-server/rtsp-stream.c:
4643 stream: Fix randomly missing streams from SDP with dynamic elements
4644 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4645 "pad-added" signal. In that case priv->srcpad could already have its caps,
4646 and they'll be sent to priv->send_src[0] pad. That means that when it
4647 connects "notify::caps" signal, that pad could already have received its
4648 caps and the signal won't be emitted anymore.
4649 In that case priv->caps stay to NULL and when building the SDP that stream
4650 gets ignored. Leading to missing video or audio when playing in client side.
4651 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4653 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4656 meson: update version
4658 === release 1.9.90 ===
4660 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4666 * gst-rtsp-server.doap:
4669 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4671 * gst/rtsp-server/rtsp-media-factory.c:
4672 * gst/rtsp-server/rtsp-media.c:
4673 * gst/rtsp-server/rtsp-stream.c:
4674 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4675 To prevent any possibly confusion with IPs or anything else.
4676 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4678 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4680 * gst/rtsp-server/rtsp-media-factory.c:
4681 * gst/rtsp-server/rtsp-media.c:
4682 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4683 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4685 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4688 configure: Depend on gstreamer 1.9.2.1
4690 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4694 Automatic update of common submodule
4695 From b18d820 to f980fd9
4697 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4701 Automatic update of common submodule
4702 From 6f2d209 to b18d820
4704 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4706 * gst/rtsp-server/rtsp-stream.c:
4707 rtsp-stream: Remove unused _locked() variant of a function
4708 It was added during refactoring.
4710 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4712 * gst/rtsp-server/rtsp-stream.c:
4713 stream: cosmetic cleanup
4714 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4716 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4718 * gst/rtsp-server/rtsp-stream.c:
4719 stream: Compare IP addresses case insensitive in more places
4720 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4722 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4725 * gst/rtsp-server/rtsp-stream.c:
4726 stream: Fix leaked joined_bin
4727 There is no need to keep a strong ref on it, and _leave_bin() was
4728 setting it to NULL before calling g_clear_object() so it was leaked.
4729 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4731 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4733 * gst/rtsp-server/rtsp-stream.c:
4734 rtsp-stream: Compare IP address strings case insensitive
4735 Otherwise IPv6 addresses might fail this comparision.
4737 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
4739 * gst/rtsp-server/rtsp-stream.c:
4740 rtsp-stream: Bind multicast sockets to ANY as before
4741 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
4743 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
4745 * gst/rtsp-server/rtsp-session.c:
4746 rtsp-session: Fix segfault when doing keep-alive after removing the session
4747 If keep-alive happens after removing the session but before finalizing the
4748 stream transport, we would segfault.
4749 https://bugzilla.gnome.org/show_bug.cgi?id=750544
4751 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
4753 * gst/rtsp-server/rtsp-stream.c:
4754 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
4755 Adding them later will cause deadlocks due to
4756 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
4757 2) adding the multicast sink
4758 3) waiting for it to get data to preroll again
4759 3) never happens because the queues after the tee are full.
4761 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
4763 * gst/rtsp-server/rtsp-stream.c:
4764 rtsp-stream: Fix up various multicast related issues
4766 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
4768 * tests/check/gst/stream.c:
4769 tests: Fix compilation
4771 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4773 * gst/rtsp-server/rtsp-client.c:
4774 * gst/rtsp-server/rtsp-stream.c:
4775 * tests/check/gst/stream.c:
4776 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
4777 This is basically reverting changes introduced in commit f62a9a7,
4778 because it was introducing various regressions:
4779 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
4780 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
4781 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
4782 - If a mcast client connects, it creates a new socket in SETUP to try to respect
4783 the destination/port given by the client in the transport, and overrides the
4784 socket already set on the udpsink element. That means that if we already had a
4785 client connected, the source address on the udp packets it receives suddenly
4787 - If a 2nd mcast client connects, the destination/port in its transport is
4788 ignored but its transport wasn't updated.
4789 What this patch does:
4790 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
4791 - Always have a tee+queue when udp is enabled. This could be optimized
4792 again in a later patch, but is more complicated. If no unicast clients
4793 connects then those elements are useless, this could be also optimized
4795 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
4796 seperated from those for unicast clients. Since we already support only
4797 one mcast address, we also create only one set of elements.
4798 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4800 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4802 * gst/rtsp-server/rtsp-stream.c:
4803 stream: factor our plug_src function
4804 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4806 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4808 * gst/rtsp-server/rtsp-stream.c:
4809 stream: factor out plug_sink function
4810 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4812 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4814 * gst/rtsp-server/rtsp-stream.c:
4815 stream: small documentation clarification
4816 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4818 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4820 * gst/rtsp-server/rtsp-stream.c:
4821 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
4822 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4824 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4826 * gst/rtsp-server/rtsp-stream.c:
4827 stream: Keep a ref on joined bin
4828 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4830 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4832 * gst/rtsp-server/rtsp-stream.c:
4833 stream: code cleanup
4834 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4836 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4838 * gst/rtsp-server/rtsp-stream.c:
4839 stream: small fix in error code path
4840 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4842 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4844 * gst/rtsp-server/rtsp-stream.c:
4845 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
4846 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
4847 but keeps unit tests.
4848 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4850 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
4855 === release 1.9.2 ===
4857 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
4863 * gst-rtsp-server.doap:
4866 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
4869 * examples/meson.build:
4871 * gst/rtsp-server/meson.build:
4872 * gst/rtsp-sink/meson.build:
4874 * pkgconfig/meson.build:
4875 * tests/check/meson.build:
4876 * tests/meson.build:
4877 Add support for Meson as alternative/parallel build system
4878 https://github.com/mesonbuild/meson
4880 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
4883 * tests/check/Makefile.am:
4884 build: silence error about pthread for 'make check' in osx
4885 Fixes "clang: error: argument unused during compilation: '-pthread'"
4887 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
4889 * gst/rtsp-server/rtsp-client.c:
4890 rtsp-client: Fix leaking of media in error cases
4891 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
4892 and myself to make the media refcounting a bit easier to follow.
4893 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4895 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4897 * gst/rtsp-server/rtsp-client.c:
4898 rtsp-client: Fix leaking of session in error cases
4899 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4901 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
4904 Automatic update of common submodule
4905 From f363b32 to f49c55e
4907 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
4912 === release 1.9.1 ===
4914 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4920 * gst-rtsp-server.doap:
4923 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4926 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
4927 https://bugzilla.gnome.org/show_bug.cgi?id=767463
4929 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4932 Automatic update of common submodule
4933 From ac2f647 to f363b32
4935 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4937 * gst/rtsp-server/rtsp-sdp.c:
4938 * gst/rtsp-server/rtsp-sdp.h:
4939 * gst/rtsp-server/rtsp-stream.c:
4940 * gst/rtsp-server/rtsp-stream.h:
4941 sdp: add rollover counters for all sender SSRC
4942 We add different crypto sessions in MIKEY, one for each sender
4943 SSRC. Currently, all of them will have the same security policy, 0.
4944 The rollover counters are obtained from the srtpenc element using the
4946 https://bugzilla.gnome.org/show_bug.cgi?id=730539
4948 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
4950 * gst/rtsp-server/rtsp-media-factory.h:
4951 * gst/rtsp-server/rtsp-server.h:
4952 docs: fix some typos
4954 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
4956 * gst/rtsp-server/Makefile.am:
4957 g-i: pass compiler env to g-ir-scanner
4958 It's what introspection.mak does as well. Should
4959 fix spurious build failures on gnome-continuous
4960 (caused by g-ir-scanner getting compiler details
4961 via python which is broken in some environments
4962 so passing the compiler details bypasses that).
4964 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
4966 * gst/rtsp-server/rtsp-session.c:
4967 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
4968 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
4969 https://bugzilla.gnome.org/show_bug.cgi?id=766619
4971 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
4973 * gst/rtsp-sink/gstrtspclientsink.c:
4974 rtspclientsink: Check return value of sscanf
4975 And just make sure we always have 0/0 if we have an error
4978 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
4980 * gst/rtsp-server/rtsp-stream.c:
4981 * tests/check/gst/rtspserver.c:
4982 * tests/check/gst/stream.c:
4983 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
4984 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
4985 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
4986 - Create unit test for shared media.
4987 https://bugzilla.gnome.org/show_bug.cgi?id=764744
4989 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4991 * gst/rtsp-server/rtsp-stream.c:
4992 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
4993 For IPv6 addresses, binding to a multicast group does not work on Linux
4994 either. Always bind to ANY and then later join the multicast group.
4995 https://bugzilla.gnome.org/show_bug.cgi?id=764679
4997 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
5000 Automatic update of common submodule
5001 From 6f2d209 to ac2f647
5003 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
5005 * gst/rtsp-server/rtsp-thread-pool.c:
5006 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
5007 Clarified why it is necessary to add source information to
5008 GstRTSPThreadImpl. See the reported bug in GLib:
5009 https://bugzilla.gnome.org/show_bug.cgi?id=720186
5010 for more information.
5011 https://bugzilla.gnome.org/show_bug.cgi?id=761702
5013 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
5015 * examples/Makefile.am:
5016 examples: Clean up CFLAGS/LDADD even more
5017 The internal .la should come first and is part of LDADD, as is
5020 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5022 * examples/Makefile.am:
5023 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5025 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5027 * gst/rtsp-server/Makefile.am:
5028 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5030 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5032 * gst/rtsp-server/rtsp-client.c:
5033 * gst/rtsp-server/rtsp-media-factory.c:
5034 * gst/rtsp-server/rtsp-media-factory.h:
5035 * gst/rtsp-server/rtsp-media.c:
5036 * gst/rtsp-server/rtsp-media.h:
5037 * gst/rtsp-server/rtsp-sdp.c:
5038 * gst/rtsp-server/rtsp-stream.c:
5039 * gst/rtsp-server/rtsp-stream.h:
5040 rtsp-server: Implement clock signalling according to RFC7273
5041 For NTP and PTP clocks we signal the actual clock that is used and signal
5042 the direct media clock offset.
5043 For all other clocks we at least signal that it's the local sender clock.
5044 This allows receivers to know which clock was used to generate the media and
5045 its RTP timestamps. Receivers can then implement network synchronization,
5046 either absolute or at least relative by getting the sender clock rate directly
5047 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5049 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5051 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5053 * gst/rtsp-sink/gstrtspclientsink.c:
5054 rtspclientsink: Add support for setting the multicast interface
5055 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5057 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5059 * gst/rtsp-server/rtsp-media-factory.c:
5060 * gst/rtsp-server/rtsp-media-factory.h:
5061 * gst/rtsp-server/rtsp-media.c:
5062 * gst/rtsp-server/rtsp-media.h:
5063 * gst/rtsp-server/rtsp-stream.c:
5064 * gst/rtsp-server/rtsp-stream.h:
5065 rtsp-media: Add support for setting the multicast interface
5066 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5068 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5070 * gst/rtsp-sink/gstrtspclientsink.c:
5071 rtspclientsink: use new gst_element_class_add_static_pad_template()
5072 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5074 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5079 === release 1.8.0 ===
5081 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5087 * gst-rtsp-server.doap:
5090 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5092 * gst/rtsp-server/rtsp-stream.c:
5093 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5094 This would get us NO_PREROLL in the bin again and break seeking.
5095 Thanks to Carlos Rafael Giani for helping to debug this!
5096 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5098 === release 1.7.91 ===
5100 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5106 * gst-rtsp-server.doap:
5109 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5111 * gst/rtsp-server/rtsp-stream.c:
5112 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5113 Without this, RECORD pipelines are broken because
5114 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5115 added later. Previously it was there earlier and due to NO_PREROLL caused the
5116 pipeline to preroll immediately
5117 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5118 as the corresponding code previously was only for PLAY pipelines.
5119 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5121 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5123 * gst/rtsp-server/rtsp-stream.c:
5124 rtsp-stream: Fix typo in the docstring
5125 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5127 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5129 * gst/rtsp-server/rtsp-stream.c:
5130 rtsp-stream: Disable multicast loopback for all our sockets
5131 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5132 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5133 loopback setting on the socket... while udpsink does which unfortunately has
5134 no effect here on Windows but on Linux.
5135 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5137 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5139 * tests/check/gst/stream.c:
5140 stream tests: added new tests
5141 Test a case when the address pool only contains multicast addresses
5142 and the client is requesting unicast udp.
5143 Added tests for multicast ports allocation.
5144 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5146 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5148 * gst/rtsp-server/rtsp-stream.c:
5149 rtsp-stream: Only bind multicast sockets to ANY on Windows
5150 On Linux it is still needed to bind to the multicast address
5151 to filter out random other packets, while on Windows binding
5152 to multicast addresses just fails.
5154 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5156 * gst/rtsp-server/rtsp-stream.c:
5157 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5158 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5159 addresses, which is something that used to work before. For unicast addresses
5160 if the pool contains none, we just allocate them as if there is no pool at
5162 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5164 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5166 * gst/rtsp-server/rtsp-client.c:
5167 * gst/rtsp-server/rtsp-stream.c:
5168 rtsp-server: Fix indentation
5170 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5172 * gst/rtsp-server/rtsp-stream.c:
5173 rtsp-stream: Don't bind the sockets to multicast addresses
5174 This works on Linux but fails completely on Windows. You're supposed
5175 to bind to ANY and then join the multicast group.
5176 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5178 === release 1.7.90 ===
5180 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5186 * gst-rtsp-server.doap:
5189 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5192 Automatic update of common submodule
5193 From b64f03f to 6f2d209
5195 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5197 * gst/rtsp-sink/gstrtspclientsink.c:
5198 * tests/check/gst/rtspclientsink.c:
5199 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5200 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5202 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5204 * tests/check/gst/media.c:
5205 * tests/check/gst/rtspclientsink.c:
5206 * tests/check/gst/rtspserver.c:
5207 * tests/check/gst/stream.c:
5208 tests: unit test fixes
5209 Removed port allocation test from the media suite.
5210 The port allocation failure is now in the stream suite.
5212 Make sure that the media is suspended after the DESCRIBE request
5213 before reconfiguring the UDP sinks.
5215 In the RECORD case we have to set async property to false
5216 for the appsink element in the test in order to make sure
5217 that the media pipeline doesn't hang in start_preroll().
5218 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5220 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5222 * gst/rtsp-server/rtsp-client.c:
5223 * gst/rtsp-server/rtsp-stream.c:
5224 * gst/rtsp-server/rtsp-stream.h:
5225 rtsp-stream: postpone UDP socket allocation until SETUP
5226 Postpone the allocation of the UDP sockets until we know
5227 what transport has been chosen by the client.
5228 Both unicast and multicast UDP sources are created in one
5230 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5232 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5234 * gst/rtsp-server/rtsp-stream.c:
5235 rtsp-stream: postpone the creation of the UDP sources
5236 Code refactoring: allocate the UDP ports after the sender and
5237 the reciver parts have been created.
5238 We postpone the creation of the UDP sources until the UDP
5239 ports have been allocated.
5240 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5242 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5244 * gst/rtsp-server/rtsp-stream.c:
5245 rtsp-stream: added function for setting UDP sources to PLAYING state
5246 Code refactoring: Introduced a function for setting UDP sources
5248 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5250 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5252 * gst/rtsp-server/rtsp-stream.c:
5253 rtsp-stream: added function for creating and configuring UDP sources
5254 Code refactoring: create and configure UDP sources in a separate function.
5255 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5257 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5259 * gst/rtsp-server/rtsp-stream.c:
5260 rtsp-stream: added function for RTP/RTCP socket configuration
5261 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5262 in a separate function.
5263 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5265 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5267 * gst/rtsp-server/rtsp-stream.c:
5268 rtsp-stream: added function for creating and configuring UDP sinks
5269 Code refactoring: create and configure UDP sinks in a separate function.
5270 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5272 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5274 * gst/rtsp-server/rtsp-stream.c:
5275 rtsp-stream: added helper function for creating the sender/receiver parts
5276 Code refactoring: introduced helper function for creating
5277 the receiver and the sender parts of the streaming pipeline.
5278 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5280 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5285 === release 1.7.2 ===
5287 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5293 * gst-rtsp-server.doap:
5296 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5298 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5299 uninstalled.pc: add support for non libtool build systems
5300 Currently the .la path is provided which requires to use libtool as
5301 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5302 It is fine as long as the application is built using libtool.
5303 So currently it is not possible to compile a GStreamer application
5304 within gst-uninstalled with CMake or other build system different
5306 This patch allows to do the following in gst-uninstalled env:
5307 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5308 gstreamer-rtsp-server-1.0)
5309 Previously it required to prepend libtool --mode=link
5310 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5312 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5314 * gst/rtsp-sink/gstrtspclientsink.c:
5315 rtspclientsink: remove check for impossible condition
5316 Goto error label checks stream to see if it needs to be unreferenced before
5317 returning, but this goto jumps happens before the stream is ever set, so it
5318 will always be NULL in this error label.
5321 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5323 * gst/rtsp-sink/gstrtspclientsink.c:
5324 rtspclientsink: clean switch statements
5325 Coverity demands for fallthrough statements to be clearly commented,
5326 to distinguish from accidental fall throughs. And it also needs all
5327 cases to finish with a break, even if the break is never going to be
5328 executed like in the case of a continue jump.
5332 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5334 * tests/check/Makefile.am:
5335 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5336 To get the CK_DEFAULT_TIMEOUT defined for all tests
5337 Also removes a 120 seconds timeout that was set as default
5338 explicitly in this module
5339 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5341 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5345 Automatic update of common submodule
5346 From 86e4663 to b64f03f
5348 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5350 * gst/rtsp-server/rtsp-media.c:
5351 rtsp-media: fix state_lock not locked again when preroll fails
5352 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5354 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5357 configure: Move plugin specific flags below all the others
5358 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5359 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5361 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5363 * gst/rtsp-sink/gstrtspclientsink.c:
5364 rtspclientsink: Simplify slightly using new -base API
5365 Use the new Mikey and SDP API in the base plugins libs
5366 to simplify some code.
5367 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5369 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5374 * gst/rtsp-sink/Makefile.am:
5375 * gst/rtsp-sink/gstrtspclientsink.c:
5376 * gst/rtsp-sink/gstrtspclientsink.h:
5377 * gst/rtsp-sink/plugin.c:
5378 * tests/check/Makefile.am:
5379 * tests/check/gst/rtspclientsink.c:
5380 rtspsink: Add rtspclientsink element
5381 Add an rtspclientsink element that accepts streams for which
5382 there is a registered payloader and sends them to
5383 an RTSP server using RECORD.
5384 Sending is synchronised to the pipeline clock. Payload-types
5385 are automatically selected. The 'new-payloader' signal is fired
5386 for custom configuration of payloaders when they are created.
5387 Can now stream a movie like this:
5389 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5390 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5392 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5393 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5394 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5396 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5398 * gst/rtsp-server/rtsp-stream.c:
5399 * gst/rtsp-server/rtsp-stream.h:
5400 rtsp-stream: Add functions for using rtsp-stream from the client
5401 Add a boolean to indicate that the rtsp-stream is running on the
5402 'client' side of an RTSP connection, for sending streams via
5403 RECORD. In that case, the roles of the client/server ports
5404 in transport setup are swapped.
5405 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5407 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5409 * gst/rtsp-server/rtsp-sdp.c:
5410 * gst/rtsp-server/rtsp-sdp.h:
5411 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5412 A new function that adds info from a GstRTSPStream into an SDP message.
5413 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5415 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5417 * gst/rtsp-server/rtsp-media.c:
5418 rtsp-media: Fix mutex beeing unlocked while they should be locked
5419 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5421 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5423 * gst/rtsp-server/rtsp-media-factory.c:
5424 rtsp-media-factory: add missing break in "clock" property setter
5427 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5429 * gst/rtsp-server/rtsp-stream.c:
5430 rtsp-stream: fixed assert during update transport
5431 When RTSP server trying update transport during multicast, it throws an
5432 assert. The assert is thrown because it is trying to get the parent of
5433 an non-existing funnel element.
5434 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5436 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5438 * gst/rtsp-server/rtsp-permissions.h:
5439 * gst/rtsp-server/rtsp-thread-pool.h:
5440 * gst/rtsp-server/rtsp-token.h:
5441 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5442 gtk-doc can handle static inline functions just fine these days,
5443 there's no need for this stuff any more.
5445 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5447 * gst/rtsp-server/rtsp-media.c:
5448 * gst/rtsp-server/rtsp-sdp.c:
5449 sdp: replace duplicated codes to call new base sdp apis
5450 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5452 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5454 * examples/test-netclock.c:
5455 test-netclock: Use the new API to configure a clock directly
5457 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5459 * gst/rtsp-server/rtsp-media-factory.c:
5460 * gst/rtsp-server/rtsp-media-factory.h:
5461 * gst/rtsp-server/rtsp-media.c:
5462 * gst/rtsp-server/rtsp-media.h:
5463 rtsp-media: Add API to directly configure a clock on the media pipelines
5465 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5467 * gst/rtsp-server/rtsp-media.c:
5468 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5470 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5472 * gst/rtsp-server/rtsp-media-factory.c:
5473 rtsp-media-factory: Add FIXME for 2.0
5475 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5477 * gst/rtsp-server/rtsp-stream.c:
5478 rtsp-stream: Fix indentation
5480 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5482 * gst/rtsp-server/rtsp-media.c:
5483 rtsp-media: Do not prepare media after media times out
5484 Deferred calls to start_prepare() can be deferred past the point until
5485 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5486 prepared to wait. Previously there was no lock and no check for this
5487 situation. This meant that a media could be prepared and unprepared
5488 simultaneously by two different threads. Now a lock is in place and a
5489 suitable check is done.
5490 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5492 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5494 * gst/rtsp-server/rtsp-client.c:
5495 * gst/rtsp-server/rtsp-media-factory.c:
5496 * gst/rtsp-server/rtsp-media-factory.h:
5497 * gst/rtsp-server/rtsp-media.c:
5498 * gst/rtsp-server/rtsp-media.h:
5499 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5500 Without TEARDOWN it might be desireable to keep the media running and continue
5501 sending data to the client, even if the RTSP connection itself is
5503 Only do this for session medias that have only UDP transports. If there's at
5504 least on TCP transport, it will stop working and cause problems when the
5505 connection is disconnected.
5506 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5508 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5513 === release 1.7.1 ===
5515 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5521 * gst-rtsp-server.doap:
5524 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5527 configure: Make -Bsymbolic check work with clang.
5528 Update the -Bsymbolic check with the version glib has. This version
5530 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5532 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5534 * gst/rtsp-server/rtsp-session-pool.c:
5535 rtsp-session-pool: Avoid dollar sign ($) in session ids
5536 Live555 in VLC strips off dollar signs and then gets very confused,
5537 we don't loose too much entropy by just skipping it.
5539 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5541 * gst/rtsp-server/rtsp-address-pool.h:
5542 * gst/rtsp-server/rtsp-auth.h:
5543 * gst/rtsp-server/rtsp-client.h:
5544 * gst/rtsp-server/rtsp-media-factory-uri.h:
5545 * gst/rtsp-server/rtsp-media-factory.h:
5546 * gst/rtsp-server/rtsp-media.h:
5547 * gst/rtsp-server/rtsp-mount-points.h:
5548 * gst/rtsp-server/rtsp-permissions.h:
5549 * gst/rtsp-server/rtsp-server.h:
5550 * gst/rtsp-server/rtsp-session-media.h:
5551 * gst/rtsp-server/rtsp-session-pool.h:
5552 * gst/rtsp-server/rtsp-session.h:
5553 * gst/rtsp-server/rtsp-stream-transport.h:
5554 * gst/rtsp-server/rtsp-stream.h:
5555 * gst/rtsp-server/rtsp-thread-pool.h:
5556 * gst/rtsp-server/rtsp-token.h:
5557 rtsp-server: Add g_autoptr() support to all types
5558 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5560 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5562 * gst/rtsp-server/rtsp-stream.c:
5563 rtsp-stream: fixed valgrind error
5564 Fixed the valgrind error in unit test. The UDP source created during
5565 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5567 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5569 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5573 Automatic update of common submodule
5574 From b319909 to 86e4663
5576 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5578 * gst/rtsp-server/rtsp-client.c:
5579 rtsp-client: suspend media during setup request
5580 SETUP request from clients needs to suspend the media to clear the
5581 prerolled buffers. Otherwise it will not affect the prerolled buffer
5582 and the prerolled buffers will be incorrect (for example block-size
5583 from setup request will not affect the prerolled buffer unless the
5584 media is suspended).
5585 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5587 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5589 * gst/rtsp-server/rtsp-stream.c:
5590 rtsp-stream: create stream pipeline based on transport
5591 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5592 only UDP is set as the transport protocol, it will not add the extra tee
5593 or queue element to the pipeline. Both these elements will be added, if
5594 it supports both TCP and UDP protocols. This improves the pipeline
5595 performance when one protocol is present.
5596 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5598 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5600 * gst/rtsp-server/rtsp-stream.c:
5601 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5602 Adding them when not needed will start some logic inside rtpbin that might be
5603 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5604 would start up a rtpjitterbuffer and behave in weird ways.
5605 We still set up the UDP sources for RTP receiving for a sender media to be
5606 able to receive any packets sent by the client for NAT traversal. They will
5607 all go to a fakesink though.
5608 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5609 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5610 receive ASYNC_DONE after a seek.
5611 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5613 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5615 * gst/rtsp-server/rtsp-stream.c:
5616 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5617 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5618 Previously we were only setting this for sender sockets, which caused looped
5619 back packets to be received on Windows if a multicast transport was used.
5621 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5623 * examples/test-record-auth.c:
5624 * examples/test-record.c:
5625 examples: Actually use the provided port in the record examples
5627 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5629 * examples/test-record-auth.c:
5630 test-record-auth: Add the option to build in TLS support
5632 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5634 * examples/test-auth.c:
5635 test-auth: Use an 'anonymous' user for unauthenticated default
5636 There's a comment on one of the resources that 'user' and 'admin'
5637 shouldn't even be able to see it, but they can if the default
5638 token is 'admin2', since that gives them access anyway.
5640 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5642 * examples/.gitignore:
5643 * examples/Makefile.am:
5644 * examples/test-record-auth.c:
5645 Add test-record-auth example
5647 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5649 * gst/rtsp-server/rtsp-client.c:
5650 * tests/check/gst/client.c:
5651 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5653 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5655 * gst/rtsp-server/rtsp-server.c:
5656 rtsp-server: Change the logic so we don't pop a NULL context
5657 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5658 will sometimes fail. This call is made before any context is pushed
5659 resulting in an attempt to pop a NULL context.
5660 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5662 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5664 * tests/check/gst/rtspserver.c:
5665 rtspserver: Add udp-mcast transport SETUP test
5666 Refactor utility functions in the test file so they can handle
5667 more than UDP and TCP as lower transport.
5668 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5670 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5672 * gst/rtsp-server/rtsp-stream.c:
5673 rtsp-stream: Always unref return value of gst_object_get_parent()
5674 Fixes a leak of a GstBin in the udp-mcast case.
5675 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5677 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5680 Automatic update of common submodule
5681 From b99800a to b319909
5683 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5686 Use new GST_ENABLE_EXTRA_CHECKS #define
5687 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5689 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5692 Automatic update of common submodule
5693 From 6babecd to b99800a
5695 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5698 Update GLib dependency to 2.40.0
5700 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5702 * examples/test-mp4.c:
5703 * gst/rtsp-server/rtsp-stream.c:
5704 stream: listen to sender ssrc signals
5705 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5707 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5710 common: update for new suppression
5711 Makes check-valgrind pass with glib 2.46
5713 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5715 * gst/rtsp-server/rtsp-media.c:
5716 rtsp-media: Take reference to media that will be prepared
5717 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5718 Later on a g_idle_source_new() is created and a pointer to the media
5719 object is passed as user data. If the media is freed before the idle
5720 source is dispatched the media object pointer is invalid, but the idle
5721 source callback expects it to still be valid. To fix this a reference to
5722 the media object is taken when registering the source callback function
5723 and a corresponding release of the reference is done when the souce is
5725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5727 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5729 * examples/test-launch.c:
5730 * examples/test-mp4.c:
5731 * examples/test-ogg.c:
5732 * examples/test-record.c:
5733 * examples/test-uri.c:
5734 rtsp-server: Fix memory leaks when context parse fails
5735 When g_option_context_parse fails, context and error variables are not getting free'd
5736 which results in memory leaks. Free'ing the same.
5737 And replacing g_error_free with g_clear_error, which checks if the error being passed
5738 is not NULL and sets the variable to NULL on free'ing.
5739 https://bugzilla.gnome.org/show_bug.cgi?id=753863
5741 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5746 === release 1.6.0 ===
5748 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5754 * gst-rtsp-server.doap:
5757 === release 1.5.91 ===
5759 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
5765 * gst-rtsp-server.doap:
5768 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
5770 * docs/libs/gst-rtsp-server-sections.txt:
5771 * gst/rtsp-server/rtsp-stream.c:
5772 stream: fix docs for recently-added get/set_buffer_size API
5773 https://bugzilla.gnome.org/show_bug.cgi?id=749095
5775 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
5777 * gst/rtsp-server/rtsp-media.c:
5778 rtsp-media: Don't crash on encrypted RTX SDP
5779 In parse_keymgmt(), don't mutate the input string that's been passed
5780 as const, especially since we might need the original value again if
5781 the same key info applies to multiple streams (RTX, for example).
5782 https://bugzilla.gnome.org/show_bug.cgi?id=754753
5784 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
5786 * examples/test-mp4.c:
5787 test-mp4: Support filenames with spaces in them. Error out on too few arguments
5789 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
5791 * examples/test-record.c:
5792 test-record: Check parameter count and print out help
5793 If no launch pipeline was supplied, print out some help
5795 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
5797 * gst/rtsp-server/rtsp-media.c:
5798 * gst/rtsp-server/rtsp-stream.c:
5799 * gst/rtsp-server/rtsp-stream.h:
5800 rtsp-stream: Implement UDP buffer size setting.
5801 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
5803 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
5804 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
5806 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
5808 * gst/rtsp-server/rtsp-media.h:
5809 rtsp-media: Fix small typo causing gtk-doc to complain
5811 === release 1.5.90 ===
5813 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5819 * gst-rtsp-server.doap:
5822 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5824 * gst/rtsp-server/rtsp-media-factory.c:
5825 media-factory: get port number through gst_rtsp_url_get_port
5826 https://bugzilla.gnome.org/show_bug.cgi?id=753473
5828 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
5830 * tests/check/gst/media.c:
5831 media-test: Removing unnecessary assertion
5832 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5834 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5836 * gst/rtsp-server/rtsp-server.c:
5837 Document that source keeps a ref on server until it's destroyed
5838 https://bugzilla.gnome.org/show_bug.cgi?id=749227
5840 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5842 * tests/check/gst/media.c:
5843 media-test: Test for multiple dynamic payload
5844 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5846 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5848 * gst/rtsp-server/rtsp-media.c:
5849 media: Only add fakesink once per pipeline
5850 The intention is to prevent going PLAYING state before pads are created.
5851 If there was mutilple dynamic payload, it would leak few fakesink and
5852 actually prevent from ever reaching playing state.
5853 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5855 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5857 * gst/rtsp-server/rtsp-media.c:
5858 Revert "rtsp-media: Only add 1 fakesink per pipeline"
5859 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
5861 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5863 * gst/rtsp-server/rtsp-media.c:
5864 rtsp-media: Only add 1 fakesink per pipeline
5865 There should be only one fakesink per pipeline, not per dynpay. This
5866 would lead to element naming clash.
5868 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
5870 * gst/rtsp-server/rtsp-media.c:
5871 rtsp-media: assertion error due to wrong condition check
5872 In media to caps function, reserved_keys array is being used for variable i,
5873 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
5874 changed it to variable j
5875 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5877 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
5879 * gst/rtsp-server/rtsp-media.c:
5880 rtsp-media: Strip keys from the fmtp that we use internally in our caps
5881 Skip keys from the fmtp, which we already use ourselves for the
5882 caps. Some software is adding random things like clock-rate into
5883 the fmtp, and we would otherwise here set a string-typed clock-rate
5884 in the caps... and thus fail to create valid RTP caps
5885 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5887 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5889 * gst/rtsp-server/rtsp-thread-pool.c:
5890 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
5891 https://bugzilla.gnome.org/show_bug.cgi?id=752640
5893 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
5896 Automatic update of common submodule
5897 From f74b2df to 9aed1d7
5899 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
5904 === release 1.5.2 ===
5906 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
5912 * gst-rtsp-server.doap:
5915 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
5917 * gst/rtsp-server/rtsp-client.c:
5918 * gst/rtsp-server/rtsp-client.h:
5919 * tests/check/gst/client.c:
5920 rtsp-client: allow application to decide what requirements are supported
5921 Add "check-requirements" signal and vfunc to allow application
5922 (and subclasses) to check the requirements.
5923 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
5924 https://bugzilla.gnome.org/show_bug.cgi?id=749417
5926 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5929 Automatic update of common submodule
5930 From 6015d26 to f74b2df
5932 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5934 * gst/rtsp-server/rtsp-media.c:
5935 rtsp-media: Always use real payloader when creating streams
5936 A bin that contains the real payloader might be used as payloader. In this
5937 case we have to get the real payloader for the various properties it provides.
5938 Example use cases for this are bins that payload some media and then have
5939 additional elements that add metadata or RTP extension headers to the stream.
5940 https://bugzilla.gnome.org/show_bug.cgi?id=750800
5942 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5944 * examples/test-netclock-client.c:
5945 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
5947 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
5949 * examples/test-netclock-client.c:
5950 * examples/test-netclock.c:
5951 test-netclock: Use new ntp-time-source property on rtpbin
5952 Select the clock time to be used as NTP time source. This allows proper
5953 synchronization between receivers, independent of sharing base times, and just
5954 requires them to use the same clock.
5956 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5958 * examples/test-netclock-client.c:
5959 * examples/test-netclock.c:
5960 test-netclock: Setting the same base time on sender and receiver is not necessary
5961 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
5963 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5965 * gst/rtsp-server/rtsp-stream.c:
5966 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
5967 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5969 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5971 * docs/libs/gst-rtsp-server.types:
5972 docs: add missing types
5973 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5975 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5977 * docs/libs/gst-rtsp-server-sections.txt:
5978 docs: add missing apis
5979 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5981 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5983 * examples/test-netclock-client.c:
5984 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
5986 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5988 * docs/libs/gst-rtsp-server-sections.txt:
5989 * gst/rtsp-server/rtsp-auth.c:
5990 * gst/rtsp-server/rtsp-auth.h:
5991 GstRTSPAuth: Add client certificate authentication support
5992 https://bugzilla.gnome.org/show_bug.cgi?id=750471
5994 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5996 * examples/test-netclock-client.c:
5997 test-netclock-client: Use new GstClock API to wait for clock synchronization
5999 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
6001 * examples/test-netclock-client.c:
6002 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
6003 A mainloop is needed to get glimagesink to display something on OSX, and
6004 the source-setup signal just makes things a little bit easier.
6006 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
6009 Automatic update of common submodule
6010 From d9a3353 to 6015d26
6012 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
6015 Automatic update of common submodule
6016 From d37af32 to d9a3353
6018 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6021 Automatic update of common submodule
6022 From 21ba2e5 to d37af32
6024 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6027 Automatic update of common submodule
6028 From c408583 to 21ba2e5
6030 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6032 * docs/libs/Makefile.am:
6033 docs: remove variables that we define in the snippet from common
6034 This is syncing our Makefile.am with upstream gtkdoc.
6036 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6039 Automatic update of common submodule
6040 From 44a3517 to c408583
6042 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6047 === release 1.5.1 ===
6049 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6055 * gst-rtsp-server.doap:
6058 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6060 * gst/rtsp-server/rtsp-client.c:
6061 rtsp-client: No flush during Teardown.
6062 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6063 backlog is empty it can happen that just a part of a message will be
6064 sent and rest is in backlog queue. If then flush during teardown
6065 just a part of message will be sent.This can lead to client miss
6066 teardown response since it expect to get the last part of message.
6067 The flushing during teardown was introduced to fix a deadlock that now
6068 is fixed more generally in handle_request by temporary setting backlog
6070 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6072 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6074 * tests/check/Makefile.am:
6075 tests: Use AM_TESTS_ENVIRONMENT
6076 Needed by the new automake test runner and the
6077 current version of the common submodule.
6079 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6081 * gst/rtsp-server/rtsp-media.h:
6082 * gst/rtsp-server/rtsp-stream.h:
6083 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6085 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6087 * gst/rtsp-server/rtsp-media.c:
6088 rtsp-media: Mark some more functions static
6090 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6092 * gst/rtsp-server/rtsp-media.c:
6093 rtsp-media: Only unblock the media in suspend() when actually changing the state
6094 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6096 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6098 * examples/test-video-rtx.c:
6099 examples: Use AVPF profile for the RTX example
6101 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6103 * gst/rtsp-server/rtsp-sdp.c:
6104 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6106 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6108 * gst/rtsp-server/rtsp-stream.c:
6109 rtsp-stream: get valid clock-rate from last-sample
6110 clock-rate in last-sample's caps is integer, not unsigned.
6111 To get this value properly, variable needs to be type-casted to int.
6112 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6114 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6118 autogen.sh: only run autopoint if gettext requested in configure.ac
6119 Not just because there happens to be a po directory.
6120 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6122 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6125 Revert "configure.ac: uncomment gettext version setup"
6126 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6127 We don't need a gettext setup here and there's no po
6128 directory either, so no reason why autopoint would be
6129 run in the first place.
6130 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6132 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6134 * examples/test-multicast.c:
6135 * examples/test-multicast2.c:
6136 * examples/test-sdp.c:
6137 * examples/test-video-rtx.c:
6138 * examples/test-video.c:
6139 * tests/test-cleanup.c:
6140 * tests/test-reuse.c:
6141 Fix timeout function signatures across tests and examples
6143 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6145 * tests/check/Makefile.am:
6146 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6147 Make sure the test environment is set up.
6148 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6150 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6153 configure: bump automake requirement to 1.14 and autoconf to 2.69
6154 This is only required for builds from git, people can still
6155 build tarballs if they only have older autotools.
6156 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6158 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6161 configure.ac: uncomment gettext version setup
6162 Fixes autogen.sh. It would run autopoint, which would complain
6163 that it could not find the gettext version in configure.ac.
6164 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6166 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6168 * examples/test-video-rtx.c:
6169 test-video-rtx: set exact payload type to PCMA payloader
6170 Setting wrong payload type causes failure to do retransmission through audio stream
6171 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6173 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6175 * gst/rtsp-server/rtsp-media.c:
6176 * gst/rtsp-server/rtsp-stream.c:
6177 * gst/rtsp-server/rtsp-stream.h:
6178 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6179 Because of duplicated g_signal_connect for request-aux-sender signal,
6180 wrong stream pointer is passed to the signal handler.
6181 Instead of passing each stream, pass stream array and get the relevant stream.
6182 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6184 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6188 Update autogen.sh to latest version from common
6189 Fixes build after aclocal_check etc. helpers have been removed.
6191 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6194 Automatic update of common submodule
6195 From bc76a8b to c8fb372
6197 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6199 * gst/rtsp-server/rtsp-stream.c:
6200 rtsp-stream: Limit the queues to 1 buffer
6201 We only need them to be able to pre-roll, queueing up more data here
6202 is only going to harm latency and memory usage.
6204 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6206 * gst/rtsp-server/rtsp-stream.c:
6207 rtsp-stream: Update comment and ASCII art to the latest code
6208 We have a queue in front of the udpsink too to prevent the pipeline from
6211 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6213 * gst/rtsp-server/rtsp-stream.c:
6214 rtsp-media: Properly return first rtptime
6215 Instead we where returning first GstBuffer timestamp. This would result
6216 in clock skew and unwanted behaviour in RTSP playback.
6217 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6219 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6221 * gst/rtsp-server/rtsp-stream.c:
6222 rtsp-stream: Don't leave buffer mapped
6223 If the seq is NULL, the RTP buffer was left mapped. We should always
6226 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6231 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6233 * gst/rtsp-server/rtsp-media-factory.c:
6234 * tests/check/gst/client.c:
6235 Fix double semicolons
6237 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6239 * gst/rtsp-server/rtsp-stream.c:
6240 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6241 This gives more accurate values than asking the payloader. There might be
6242 queueing happening between the payloader and the sink.
6243 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6245 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6247 * gst/rtsp-server/rtsp-media.c:
6248 rtsp-media: Don't seek for PLAY if the position will not change
6249 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6251 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6253 * gst/rtsp-server/rtsp-media.c:
6254 rtsp-media: Don't include payload type in the caps for framesize
6255 When the sdp media attribute framesize are converted to caps
6256 the <payload> should not be included.
6257 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6258 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6260 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6262 * gst/rtsp-server/rtsp-sdp.c:
6263 rtsp-sdp: add payload type to the sdp framesize attribute
6264 The sdp framesize attribute is desribed in RFC6064. It is specified
6265 for payloading of H263 and has the following form
6266 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6267 should be added to the caps in a payloader and the <payload type> should
6268 be added by the rtsp-server.
6269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6271 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6273 * examples/test-uri.c:
6274 examples: test-uri: fix tainted variable
6275 Insignificant but this keeps Coverity happy.
6278 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6280 * examples/.gitignore:
6281 * examples/Makefile.am:
6282 * examples/test-netclock-client.c:
6283 * examples/test-netclock.c:
6284 examples: Add a simple example of network synch for live streams.
6285 An example server and client that works for synchronising live streams
6286 only - as it can't support pause/play.
6288 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6290 * gst/rtsp-server/rtsp-media-factory.c:
6291 * gst/rtsp-server/rtsp-media-factory.h:
6292 rtsp-media-factory: Add functions to set/get the media gtype
6293 Allow specifying the GType of a GstRtspMedia subclass to create
6294 as a simpler way to get the factory to create a custom
6295 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6297 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6299 * gst/rtsp-server/rtsp-media.c:
6300 rtsp-media: fix double unlock in _get_buffer_size()
6301 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6302 because of double g_mutex_unlock () usage.
6303 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6305 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6307 * gst/rtsp-server/rtsp-session-pool.c:
6308 * gst/rtsp-server/rtsp-session.c:
6309 * gst/rtsp-server/rtsp-session.h:
6310 rtsp-session: Use monotonic time for RTSP session timeout
6311 Changed RTSP session timeout handling to monotonic time
6312 and deprecating the API for current system time.
6313 This fixes timeouts when the system time changes.
6314 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6316 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6318 * gst/rtsp-server/rtsp-client.c:
6319 * gst/rtsp-server/rtsp-media.c:
6320 rtsp-client: Only error out in PLAY if seeking actually failed
6321 If the media was just not seekable, we continue from whatever position we are
6322 and let the client decide if that is what is wanted or not.
6323 Only if the actual seek failed, we can't really recover and should error out.
6325 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6327 * gst/rtsp-server/rtsp-stream.c:
6328 rtsp-stream: Add necessary queues between tee and multiudpsink
6329 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6331 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6333 * gst/rtsp-server/rtsp-client.c:
6334 * gst/rtsp-server/rtsp-media.c:
6335 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6336 Instead error out properly the same way as if the SEEKING query already
6339 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6341 * gst/rtsp-server/rtsp-stream.h:
6342 rtsp-stream: minor code formatting fix
6344 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6346 * gst/rtsp-server/rtsp-media.c:
6347 rtsp-media: fix logic for collect_streams
6348 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6349 all streams it knows if it got any, and can check if the transport mode is OK.
6352 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6354 * gst/rtsp-server/rtsp-media.c:
6355 rtsp-media: Don't set the transport mode based on what elements we find
6356 Just print a warning if the one that was set before disagrees with what
6357 elements we found. It must already be set to something before as this
6358 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6359 and we would reject ANNOUNCE if the RECORD flag was not set.
6361 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6363 * tests/check/gst/rtspserver.c:
6364 tests: rtspserver: rename shadowed variable
6365 We have two different 'sink' variables here,
6366 rename one of them for clarity.
6368 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6370 * gst/rtsp-server/rtsp-client.c:
6371 rtsp-client: fix awkward if clause
6373 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6375 * examples/test-uri.c:
6376 examples: test-uri: improve uri argument handling and accept file names
6377 Print an error if the argument passed is not a URI and can't
6378 be converted into one, or no arguments have been provided.
6380 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6382 * examples/test-uri.c:
6383 examples: test-uri: don't remove mount point after 10 seconds
6384 It's very irritating when trying to test stuff repeatedly
6385 and serves no real purpose other than showing that it can
6388 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6390 * examples/.gitignore:
6391 examples: add new test-record to .gitignore
6393 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6395 * examples/test-record.c:
6396 * gst/rtsp-server/rtsp-client.c:
6397 * gst/rtsp-server/rtsp-media-factory.c:
6398 * gst/rtsp-server/rtsp-media-factory.h:
6399 * gst/rtsp-server/rtsp-media.c:
6400 * gst/rtsp-server/rtsp-media.h:
6401 * tests/check/gst/rtspserver.c:
6402 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6404 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6406 * examples/test-record.c:
6407 test-record: Set latency for playback-style example to 2s instead of 200ms
6409 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6411 * tests/check/gst/rtspserver.c:
6412 tests: add some unit tests for ANNOUNCE and RECORD
6413 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6415 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6417 * gst/rtsp-server/rtsp-client.c:
6418 rtsp-client: fix a couple of leaks in handle_announce
6420 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6422 * gst/rtsp-server/rtsp-media-factory.c:
6423 * gst/rtsp-server/rtsp-media-factory.h:
6424 * gst/rtsp-server/rtsp-media.c:
6425 * gst/rtsp-server/rtsp-media.h:
6426 rtsp-media: Expose latency setting for setting the rtpbin latency
6428 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6430 * examples/test-record.c:
6431 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6433 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6435 * gst/rtsp-server/rtsp-stream.c:
6436 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6438 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6440 * examples/Makefile.am:
6441 * examples/test-record.c:
6442 * gst/rtsp-server/rtsp-client.c:
6443 * gst/rtsp-server/rtsp-client.h:
6444 * gst/rtsp-server/rtsp-media-factory.c:
6445 * gst/rtsp-server/rtsp-media-factory.h:
6446 * gst/rtsp-server/rtsp-media.c:
6447 * gst/rtsp-server/rtsp-media.h:
6448 * gst/rtsp-server/rtsp-session-media.c:
6449 * gst/rtsp-server/rtsp-stream.c:
6450 * gst/rtsp-server/rtsp-stream.h:
6451 Add initial support for RECORD
6452 We currently only support media that is RECORD or PLAY only, not both at once.
6453 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6455 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6457 * gst/rtsp-server/rtsp-stream.c:
6458 rtsp-stream: RTCP and RTP transport cache cookies seperated
6459 RTCP packets were not sent because the same tr_cache_cookie was used for
6460 both RTP and RTCP. So only one of the tr_cache lists were populated
6461 depending on which one was sent first. If the tr_cache list is not
6462 populated then no packets can be sent. Most often this happened to be
6463 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6464 resulted in both the tr_cache_lists to be populated regardless of which
6466 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6468 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6470 * gst/rtsp-server/rtsp-stream.c:
6471 rtsp-stream: fix false compiler warning
6472 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6474 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6476 * gst/rtsp-server/rtsp-client.c:
6477 rtsp-client: log interleaved data received
6479 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6481 * gst/rtsp-server/rtsp-client.c:
6482 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6484 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6486 * gst/rtsp-server/rtsp-client.c:
6487 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6489 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6491 * gst/rtsp-server/rtsp-client.c:
6492 rtsp-client: Use a random session ID in the SDP
6493 RFC4566 Section 5.2 says that it should make the username, session id,
6494 nettype, addrtype and unicast address tuple globally unique. Always using
6495 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6496 Instead let's create a 64 bit random number, which at least brings us
6497 closer to the goal of global uniqueness.
6498 https://tools.ietf.org/html/rfc4566#section-5.2
6500 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6502 * examples/test-launch.c:
6503 * examples/test-mp4.c:
6504 * examples/test-ogg.c:
6505 * examples/test-uri.c:
6506 examples: Don't call gst_init() and gst_get_option_group()
6507 The latter calls the former at the appropriate time.
6509 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6511 * gst/rtsp-server/rtsp-client.c:
6512 rtsp-client: Drop trailing \0 of RTSP DATA messages
6513 We add a trailing \0 in GstRTSPConnection to make parsing of
6514 string message bodies easier (e.g. the SDP from DESCRIBE) but
6515 for actual data this means we have to drop it or otherwise
6516 create invalid data.
6518 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6520 * gst/rtsp-server/rtsp-stream.c:
6521 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6522 Fixes crash when two threads access handle_new_sample() at the same
6523 time, one for RTP, one for RTCP.
6524 Otherwise, when iterating over the transports cache, it might be modified by
6525 another thread at the same time if the transports cookie has changed.
6526 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6528 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6530 * gst/rtsp-server/rtsp-stream.c:
6531 rtsp-stream: Set format=TIME on our app sources for TCP
6533 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6535 * gst/rtsp-server/rtsp-session-pool.c:
6536 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6537 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6538 RFC 2326 states that session IDs may consist of alphanumeric as well as
6539 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6540 Previously the session ID was URI-escaped, this meant that any character
6541 which was not alphanumeric or any of the characters +-._~ would be
6542 percent encoded. While the RFC (surprisingly) mentions that linear white
6543 space in session IDs should be URI-escaped, it does not say anything
6544 about other characters. Moreover no white space is allowed in the
6545 session ID. Finally the percent character which is the result of
6546 URI-escaping is not allowed in a session ID.
6547 So there is no reason to do any URI-escaping, and now it is removed.
6548 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6550 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6553 Automatic update of common submodule
6554 From f2c6b95 to bc76a8b
6556 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6559 Fix 'make check' from top-level directory
6561 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6563 * examples/test-launch.c:
6564 * examples/test-mp4.c:
6565 * examples/test-ogg.c:
6566 * examples/test-uri.c:
6567 examples: Add command-line parsing and take a 'port' argument
6568 This allows users to run multiple servers on different ports for testing.
6569 Only done for examples that actually take arguments and hence are capable of
6570 outputting different streams for each instance on each port.
6571 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6573 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6575 * gst/rtsp-server/rtsp-client.c:
6576 * gst/rtsp-server/rtsp-client.h:
6577 rtsp-client: Add a send_message default signal handler
6578 This allows subclasses to easily hook into the response sending
6579 mechanism without doing everything from a signal, which seems
6580 awkward from subclasses.
6582 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6585 Automatic update of common submodule
6586 From ef1ffdc to f2c6b95
6588 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6592 configure: add --disable-examples switch
6593 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6595 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6597 * examples/.gitignore:
6598 * examples/Makefile.am:
6599 * examples/test-video-rtx.c:
6600 examples: add a retransmisison example implementing RFC4588
6601 Currently only SSRC-multiplexed rtx streams are supported
6603 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6605 * gst/rtsp-server/rtsp-stream.c:
6606 rtsp-stream: Fix some minor memory leaks
6608 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6610 * gst/rtsp-server/rtsp-media.c:
6611 rtsp-media: Some minor cleanup
6613 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6615 * gst/rtsp-server/rtsp-stream.c:
6616 rtsp-stream: Fix compiler warnings
6617 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6618 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6620 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6621 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6624 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6626 * docs/libs/gst-rtsp-server-sections.txt:
6627 * gst/rtsp-server/rtsp-media-factory.c:
6628 * gst/rtsp-server/rtsp-media-factory.h:
6629 * gst/rtsp-server/rtsp-media.c:
6630 * gst/rtsp-server/rtsp-media.h:
6631 * gst/rtsp-server/rtsp-sdp.c:
6632 * gst/rtsp-server/rtsp-stream.c:
6633 * gst/rtsp-server/rtsp-stream.h:
6634 media: implement ssrc-multiplexed retransmission support
6635 based off RFC 4588 and the server-rtpaux example in -good
6637 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6639 * gst/rtsp-server/rtsp-client.c:
6640 * gst/rtsp-server/rtsp-stream-transport.c:
6641 * gst/rtsp-server/rtsp-stream.c:
6642 rtsp: Ref transports in hash table.
6643 Also ref streams for transports.
6644 This solves a crash when reciving a rtcp after teardown but before
6646 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6648 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6651 Automatic update of common submodule
6652 From 7bb2bce to ef1ffdc
6654 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6656 * gst/rtsp-server/rtsp-client.c:
6657 client: refactor cleanup of cached media
6659 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6661 * tests/check/gst/client.c:
6663 The session leak is now fixed, lets remove those FIXME comments.
6665 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6667 * tests/check/gst/rtspserver.c:
6668 tests: Test to setup two sessions on one connection
6669 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6671 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6673 * tests/check/gst/rtspserver.c:
6674 tests: Test setup with tcp transport
6675 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6677 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6679 * gst/rtsp-server/rtsp-client.c:
6680 client: Configure transport after creating session media
6681 The default implementation of configure_client_transport() in
6682 rtsp-client uses the session media when it chooses channels for
6683 interleaved traffic.
6684 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6686 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6688 * gst/rtsp-server/rtsp-client.c:
6689 * gst/rtsp-server/rtsp-session-media.c:
6690 client: Stop caching media in client when doing setup
6691 If the media has been managed by a session media, it should not be
6692 cached in the client any longer. The GstRTSPSessionMedia object is now
6693 responsible for unpreparing the GstRTSPMedia object using
6694 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6696 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6698 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6700 * gst/rtsp-server/rtsp-stream.c:
6701 rtsp-stream: unref srtp decoder when leaving bin
6702 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6704 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6706 * gst/rtsp-server/rtsp-client.c:
6707 rtsp-client: mikey memory leaks
6708 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6710 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6713 Automatic update of common submodule
6714 From 84d06cd to 7bb2bce
6716 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6719 Parallelise 'make check-valgrind'
6721 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6724 Automatic update of common submodule
6725 From a8c8939 to 84d06cd
6727 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6730 Automatic update of common submodule
6731 From 36388a1 to a8c8939
6733 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6735 * gst/rtsp-server/rtsp-media.c:
6736 rtsp-media: deactivate media when shutting down from paused
6737 This was only done when going directly from playing.
6738 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
6740 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6742 * gst/rtsp-server/rtsp-client.c:
6743 * gst/rtsp-server/rtsp-context.h:
6744 rtsp-client: add stream transport to context
6745 We add the stream transport to the context so we can get the configured
6746 client stream transport in the setup request signal.
6747 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
6749 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6751 * gst/rtsp-server/rtsp-stream.c:
6752 stream: release lock even not all transports have been removed
6753 We don't want to keep the lock even we return FALSE because not all the
6754 transports have been removed. This could lead into a deadlock.
6755 https://bugzilla.gnome.org/show_bug.cgi?id=737797
6757 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
6759 * gst/rtsp-server/rtsp-sdp.c:
6760 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
6761 These were renamed in GstRTPBasePayload in 1.0
6763 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6765 * gst/rtsp-server/rtsp-client.c:
6766 client: set session media to NULL without the lock
6767 We need to set session medias to NULL without the client lock otherwise
6768 we can end up in a deadlock if another thread is waiting for the lock
6769 and media unprepare is also waiting for that thread to end.
6770 https://bugzilla.gnome.org/show_bug.cgi?id=737690
6772 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
6774 * gst/rtsp-server/rtsp-media.c:
6775 rtsp-media: Set state to UNPREPARING in all cases
6777 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
6779 * gst/rtsp-server/rtsp-media.c:
6780 media: set state to unpreparing when unprepare is initiated
6781 https://bugzilla.gnome.org/show_bug.cgi?id=737675
6783 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
6785 * gst/rtsp-server/rtsp-client.c:
6786 rtsp-client: Remove backlog limit while processings requests
6787 If the backlog limit is kept two cases of deadlocks may be
6788 encountered when streaming over TCP. Without the backlog
6789 limit this deadlocks can not happen, at the expence of
6791 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
6793 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
6795 * gst/rtsp-server/rtsp-client.c:
6796 rtsp-client: do not free main context before rtsp watch
6797 https://bugzilla.gnome.org/show_bug.cgi?id=737110
6799 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
6801 * tests/check/gst/rtspserver.c:
6802 tests: Extend unit test timeout to accomodate for valgrind
6803 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6805 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
6807 * gst/rtsp-server/rtsp-client.c:
6808 * gst/rtsp-server/rtsp-session.c:
6809 * gst/rtsp-server/rtsp-stream-transport.c:
6810 rtsp-*: Treat sending packets to clients as keepalive
6811 As long as gst-rtsp-server can successfully send RTP/RTCP data to
6812 clients then the client must be reading. This change makes the server
6813 timeout the connection if the client stops reading.
6814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6816 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
6818 * gst/rtsp-server/rtsp-client.c:
6819 rtsp-client: Allow backlog to grow while expiring session
6820 Allow the send backlog in the RTSP watch to grow to unlimited size while
6821 attempting to bring the media pipeline to NULL due to a session
6822 expiring. Without this change the appsink element cannot change state
6823 because it is blocked while rendering data in the new_sample callback.
6824 This callback will block until it has successfully put the data into the
6825 send backlog. There is a chance that the send backlog is full at this
6826 point which means that the callback may block for a long time, possibly
6827 forever. Therefore the media pipeline may also be prevented from
6828 changing state for a long time.
6829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6831 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
6833 * gst/rtsp-server/rtsp-client.c:
6834 rtsp-client: Make old compilers happy
6835 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
6836 Just in case that guint8 doesn't fit in a pointer. Just in case ...
6838 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
6840 * gst/rtsp-server/rtsp-client.c:
6841 client: raise the backlog limits before pausing
6842 We need to raise the backlog limits before pausing the pipeline or else
6843 the appsink might be blocking in the render method in wait_backlog() and
6844 we would deadlock waiting for paused.
6845 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
6847 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
6849 * gst/rtsp-server/rtsp-client.c:
6850 client: make define for the WATCH_BACKLOG
6851 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
6853 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
6855 * gst/rtsp-server/rtsp-client.c:
6856 client: simplify session transport handling
6857 link/unlink of the transport in a session was done to keep track of all
6858 TCP transports and to send RTP/RTCP data to the streams. We can simplify
6859 that by putting all the TCP transports in a hashtable indexed with the
6861 We also don't need to link/unlink the transports when we pause/resume
6862 the streams. The same effect is already achieved when we pause/play the
6863 media. Indeed, when we pause the media, the transport is removed from
6864 the media and the callbacks will not be called anymore.
6865 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
6867 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
6869 * gst/rtsp-server/rtsp-stream-transport.c:
6870 * gst/rtsp-server/rtsp-stream-transport.h:
6871 stream-transport: make method to handle received data
6872 Make a method to handle the data received on a channel. It sends the
6873 data to the stream of the transport on the RTP or RTCP pads based on
6876 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
6878 * examples/test-mp4.c:
6879 test: add example of dumping RTCP reports
6881 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
6883 * gst/rtsp-server/rtsp-media.c:
6884 * gst/rtsp-server/rtsp-stream.c:
6885 * gst/rtsp-server/rtsp-stream.h:
6886 rtsp-media: Make sure that sequence numbers are monotonic after pause
6887 The sequence number is not monotonic for RTP packets after pause. The
6888 reason is basepayloader generates a randon sequence number when the
6889 pipeline goes from ready to pause. With this fix generation of sequence
6890 number will be monotonic when going from pause to play request.
6891 https://bugzilla.gnome.org/show_bug.cgi?id=736017
6893 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
6895 * gst/rtsp-server/rtsp-client.c:
6896 rtsp-client: Protect saved clients watch with a mutex
6897 Fixes a crash when close() is called while merging clients
6898 in handle_tunnel(). In that case close() would destroy the
6899 watch while it is still being used in handle_tunnel().
6900 https://bugzilla.gnome.org/show_bug.cgi?id=735570
6902 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
6904 * gst/rtsp-server/rtsp-stream.c:
6905 rtsp-stream: Remove the multicast group udp sources when removing from the bin
6907 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6909 * gst/rtsp-server/rtsp-media.c:
6910 * gst/rtsp-server/rtsp-stream.c:
6911 * gst/rtsp-server/rtsp-stream.h:
6912 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
6913 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
6914 seeking and will always continue counting the time. This leads to
6915 the NPT after a backwards seek to be something completely different
6916 to the actual seek position.
6917 https://bugzilla.gnome.org/show_bug.cgi?id=732644
6919 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
6921 * examples/test-appsrc.c:
6922 examples: fix another reference leak
6923 gst_rtsp_media_get_element() returns a new ref.
6925 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6927 * examples/test-appsrc.c:
6928 examples: unref element after usage
6929 gst_bin_get_by_name_recurse_up() returns an element
6930 reference that must be unreffed after usage.
6931 https://bugzilla.gnome.org/show_bug.cgi?id=734546
6933 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
6935 * gst/rtsp-server/rtsp-media.c:
6936 signals: Fix copy-pasto in target-state signal offset
6938 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
6942 Makefile: Add usage of build-checks step
6943 Allows building checks without running them
6945 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
6947 * gst/rtsp-server/rtsp-stream.c:
6948 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
6949 When a UDP multicast transport is used it is expected that the server listens
6950 for RTP and RTCP packets on the multicast group with the corresponding port.
6951 Without this we will never get RTCP packets from clients in multicast mode.
6952 https://bugzilla.gnome.org/show_bug.cgi?id=732238
6954 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6959 === release 1.4.0 ===
6961 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6967 * gst-rtsp-server.doap:
6970 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
6972 * gst/rtsp-server/rtsp-media.h:
6973 media: correct misspelled words in description
6974 https://bugzilla.gnome.org/show_bug.cgi?id=733244
6976 === release 1.3.91 ===
6978 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6984 * gst-rtsp-server.doap:
6987 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
6989 * docs/libs/gst-rtsp-server-sections.txt:
6992 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
6994 * gst/rtsp-server/rtsp-server.c:
6995 server: implement client REMOVE filter
6997 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
6999 * gst/rtsp-server/rtsp-client.c:
7000 * gst/rtsp-server/rtsp-client.h:
7001 client: expose _close() method
7002 Expose a previously internal close method to close the client
7005 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
7007 * gst/rtsp-server/rtsp-session-pool.c:
7008 session-pool: signal session-removed outside of the lock
7009 Release the lock before emiting the session-removed signal.
7011 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
7013 * gst/rtsp-server/rtsp-client.c:
7014 * gst/rtsp-server/rtsp-server.c:
7015 * gst/rtsp-server/rtsp-session-pool.c:
7016 * gst/rtsp-server/rtsp-session.c:
7017 * gst/rtsp-server/rtsp-stream.c:
7018 filter: Release lock in filter functions
7019 Release the object lock before calling the filter functions. We need to
7020 keep a cookie to detect when the list changed during the filter
7021 callback. We also keep a hashtable to make sure we only call the filter
7022 function once for each object in case of concurrent modification.
7023 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7025 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7027 * gst/rtsp-server/rtsp-client.c:
7028 client: check if watch is set in handle_teardown()
7029 The unit tests run without a watch
7031 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7033 * tests/check/gst/client.c:
7034 client tests: send teardown to cleanup session
7036 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7038 * tests/check/gst/rtspserver.c:
7039 server tests: send teardown to cleanup session
7041 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7043 * gst/rtsp-server/rtsp-client.c:
7044 client: keep ref to client for the session removed handler
7045 This extra ref will be dropped when all client sessions have been
7046 removed. A session is removed when a client sends teardown, closes its
7047 endpoint of the TCP connection or the sessions expires.
7048 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7050 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7052 * gst/rtsp-server/rtsp-client.c:
7053 * gst/rtsp-server/rtsp-session.c:
7054 * tests/check/gst/client.c:
7055 client: manage media in session as a last step
7056 Once we manage a media in a session, we can't unmanage it anymore
7057 without destroying it. Therefore, first check everything before we
7058 manage the media, otherwise if something is wrong we have no way to
7060 If we created a new session and something went wrong, remove the session
7061 again. Fixes a leak in the unit test.
7063 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7065 * examples/test-mp4.c:
7066 * examples/test-ogg.c:
7067 examples: print 'stream ready at url' for mp4 and ogg example
7069 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7071 * gst/rtsp-server/rtsp-client.c:
7072 * gst/rtsp-server/rtsp-sdp.c:
7073 rtsp: fix for MIKEY api change
7075 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7077 * gst/rtsp-server/rtsp-client.c:
7078 client: free watch context only once
7079 The watch context is freed when the source is destroyed. Avoids
7080 a CRITICAL when we try to unref the context twice.
7082 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7084 * gst/rtsp-server/rtsp-client.c:
7087 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7089 * gst/rtsp-server/rtsp-client.c:
7090 client: protect sessions with lock
7091 Protect the list of sessions with the lock.
7092 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7094 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7096 * gst/rtsp-server/rtsp-client.c:
7097 Client: keep a ref to the session
7098 Don't just keep a weak ref to the session objects but use a hard ref. We
7099 will be notified when a session is removed from the pool (expired) with
7100 the new session-removed signal.
7101 Don't automatically close the RTSP connection when all the sessions of
7102 a client are removed, a client can continue to operate and it can create
7103 a new session if it wants. If you want to remove the client from the
7104 server, you have to use gst_rtsp_server_client_filter() now.
7105 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7106 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7108 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7110 * gst/rtsp-server/rtsp-session-pool.c:
7111 * gst/rtsp-server/rtsp-session-pool.h:
7112 session-pool: add session-removed signal
7113 Add a signal to be notified when a session is removed from the pool.
7115 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7117 * gst/rtsp-server/Makefile.am:
7118 * gst/rtsp-server/rtsp-server.h:
7119 Make rtsp-server.h a single-include header, use it for G-I
7120 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7122 === release 1.3.90 ===
7124 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7130 * gst-rtsp-server.doap:
7133 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7135 * gst/rtsp-server/rtsp-stream.c:
7136 stream: crypto can be NULL
7138 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7140 * gst/rtsp-server/rtsp-client.c:
7141 * gst/rtsp-server/rtsp-media.c:
7142 * gst/rtsp-server/rtsp-mount-points.c:
7143 introspection: add missing allow-none annotations
7144 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7146 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7148 * gst/rtsp-server/rtsp-address-pool.c:
7149 * gst/rtsp-server/rtsp-media.c:
7150 * gst/rtsp-server/rtsp-session-media.c:
7151 * gst/rtsp-server/rtsp-session-pool.c:
7152 * gst/rtsp-server/rtsp-stream-transport.c:
7153 * gst/rtsp-server/rtsp-stream.c:
7154 * gst/rtsp-server/rtsp-token.c:
7155 introspection: add (nullable) annotations to return values
7156 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7158 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7160 * gst/rtsp-server/rtsp-client.c:
7161 * gst/rtsp-server/rtsp-stream.c:
7162 gi: improve annotations
7163 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7165 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7167 * gst/rtsp-server/rtsp-client.c:
7168 * gst/rtsp-server/rtsp-media-factory.c:
7169 * gst/rtsp-server/rtsp-media.c:
7170 * gst/rtsp-server/rtsp-server.c:
7171 signals: use generic marshal function
7172 Use the generic C marshal function.
7173 Use more explicit type instead of G_TYPE_POINTER
7175 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7177 * gst/rtsp-server/rtsp-context.h:
7178 context: add type macro
7180 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7182 * gst/rtsp-server/rtsp-client.c:
7183 * gst/rtsp-server/rtsp-sdp.c:
7184 * gst/rtsp-server/rtsp-sdp.h:
7185 sdp: hide key length defines
7186 They don't have a namespace.
7188 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7193 === release 1.3.3 ===
7195 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7201 * gst-rtsp-server.doap:
7204 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7206 * gst/rtsp-server/rtsp-client.c:
7207 * gst/rtsp-server/rtsp-sdp.c:
7208 * gst/rtsp-server/rtsp-sdp.h:
7209 mikey: add different key length parameters
7210 Add encryption and authentication key length parameters to MIKEY. For
7211 the encoders, the key lengths are obtained from the cipher and auth
7212 algorithms set in the caps. For the decoders, they are obtained while
7213 parsing the key management from the client.
7214 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7216 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7218 * tests/check/gst/stream.c:
7219 stream tests: Make sure we get right multicast address from stream
7220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7222 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7224 * gst/rtsp-server/rtsp-client.c:
7225 client: ref the context until rtsp watch is alive
7226 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7228 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7230 * gst/rtsp-server/rtsp-client.c:
7231 client: Destroy the rtsp watch after connection close
7233 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7235 * gst/rtsp-server/rtsp-media.c:
7236 media: fix confusing comment
7238 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7240 * gst/rtsp-server/rtsp-session.c:
7241 rtsp-session: Timeout in header.
7242 Adding the possbilty to always have timout in header.
7243 This is configurabe with setting "timeout-always-visible".
7244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7246 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7251 === release 1.3.2 ===
7253 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7260 * gst-rtsp-server.doap:
7263 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7266 Automatic update of common submodule
7267 From 211fa5f to 1f5d3c3
7269 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7271 * gst/rtsp-server/rtsp-client.c:
7272 client: store TCP ports in transport
7273 Store the TCP ports in the transport when we are doing RTSP over TCP.
7274 This way, we can easily get to the ports from the transport.
7275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7277 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7279 * gst/rtsp-server/rtsp-stream.c:
7280 stream: add signals for new RTP/RTCP encoders
7281 New signals to allow the user to configure the dynamically created
7283 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7285 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7287 * gst/rtsp-server/rtsp-media.c:
7288 * gst/rtsp-server/rtsp-media.h:
7289 media: Make suspend()/unsuspend() virtual
7290 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7292 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7294 * gst/rtsp-server/rtsp-client.c:
7295 client: fix send-message signal marshaller
7296 Use generic marshalling for the send-message signal. It has
7297 two POINTER arguments, not just one.
7298 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7300 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7302 * tests/check/gst/media.c:
7303 tests: add and remove pads only once
7304 In this test we simulate a dynamic pad by watching the caps event.
7305 Because of renegotiation in the base payloader now, this caps is sent
7306 multiple times but we can only deal with 1 invocation, use a variable to
7307 only 'add and remove' the pad once.
7309 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7311 * tests/check/gst/rtspserver.c:
7312 tests: add unit test for correct handling of Require headers
7313 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7315 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7317 * gst/rtsp-server/rtsp-client.c:
7318 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7319 Servers must handle Require headers and must report a failure
7320 if they don't handle any of the Required options, see RFC 2326,
7321 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7322 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7324 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7329 === release 1.3.1 ===
7331 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7337 * gst-rtsp-server.doap:
7340 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7343 Automatic update of common submodule
7344 From bcb1518 to 211fa5f
7346 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7351 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7353 * tests/check/gst/sessionmedia.c:
7354 tests: fix memory leak in sessionmedia unit test
7356 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7358 * gst/rtsp-server/rtsp-client.c:
7359 client: emit a signal before sending a message
7360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7362 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7364 * gst/rtsp-server/rtsp-client.c:
7365 client: pass context to send_message
7366 Pass the current context to send_message, we will need it later.
7368 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7370 * gst/rtsp-server/rtsp-client.c:
7371 client: fix typo in comment
7373 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7375 * gst/rtsp-server/rtsp-media.c:
7376 media: Do not stop thread twice if default_prepare() fails
7378 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7380 * gst/rtsp-server/rtsp-client.c:
7381 client: set the watch to flushing before going to NULL
7382 First set the watch to flushing so that we unblock any current and
7383 future attempt to send data on the watch, Then set the pipeline to
7385 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7387 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7389 * gst/rtsp-server/rtsp-session-pool.c:
7390 * tests/check/gst/sessionpool.c:
7391 rtsp-session-pool: Fixes annotation
7392 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7393 in the sessionpool test.
7394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7396 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7398 * gst/rtsp-server/rtsp-media.c:
7399 * gst/rtsp-server/rtsp-media.h:
7400 media: make media_prepare virtual
7401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7403 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7405 * gst/rtsp-server/rtsp-media.c:
7406 * tests/check/gst/media.c:
7407 media: stop the thread in more error cases
7409 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7411 * gst/rtsp-server/rtsp-media.c:
7412 * tests/check/gst/media.c:
7413 media: allow NULL as the thread
7414 Use the default context whan passing a NULL thread.
7416 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7418 * gst/rtsp-server/rtsp-client.c:
7419 rtsp-client: indent cleanup
7420 Coverity was moaning about unreachable code, and I think it was just
7421 confused by { being before the label. We'll see if it pops up again.
7424 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7426 * gst/rtsp-server/rtsp-client.c:
7427 * gst/rtsp-server/rtsp-media.c:
7428 client: Add drop-backlog property
7429 When we have too many messages queued for a client (currently hardcoded
7430 to 100) we overflow and drop the messages. Add a drop-backlog property
7431 to control this behaviour. Setting this property to FALSE will retry
7432 to send the messages to the client by waiting for more room in the
7434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7436 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7438 * gst/rtsp-server/rtsp-client.c:
7439 client: support for POST before GET when setting up a tunnel
7441 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7443 * gst/rtsp-server/rtsp-client.c:
7444 client: remove watch of the second client after http tunnel setup
7445 The second client will be freed after the HTTP tunnel has been set up.
7446 Make sure it's RTSP watch is never dispatched again.
7447 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7449 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7451 * gst/rtsp-server/rtsp-media.c:
7452 * tests/check/gst/media.c:
7453 media: Make media_prepare() fail if port allocation fails
7454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7456 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7458 * tests/check/gst/media.c:
7459 media test: cleanup the thread pool in tests
7461 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7463 * gst/rtsp-server/rtsp-media.c:
7464 * tests/check/gst/media.c:
7465 rtsp-media: Unblock blocked streams in unprepare
7466 The streams will be blocked when a live media is prepared.
7467 The streams should be unblocked in gst_rtsp_media_unprepare.
7468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7470 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7472 * gst/rtsp-server/rtsp-media.c:
7473 media: release the state lock when going to NULL
7474 Set our state to UNPREPARING and release the state-lock before
7475 setting the pipeline to the NULL state. This way, any pad-added
7476 callback will be able to take the state-lock and check that we are now
7477 unpreparing instead of deadlocking.
7478 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7480 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7482 * gst/rtsp-server/rtsp-media.c:
7483 media: protect status with lock
7484 Make sure we only update the status with the lock.
7486 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7488 * gst/rtsp-server/rtsp-client.c:
7489 * gst/rtsp-server/rtsp-sdp.c:
7490 rtsp: update for MIKEY API changes
7492 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7494 * gst/rtsp-server/rtsp-client.c:
7495 client: parse the mikey response from the client
7496 Parse the mikey response from the client and update the policy for
7499 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7501 * gst/rtsp-server/rtsp-stream.c:
7502 * gst/rtsp-server/rtsp-stream.h:
7503 stream: add method to set crypto info
7504 Make a method to configure the crypto information of a stream.
7505 Set udpsrc in READY instead of PAUSED so that we can configure caps
7508 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7510 * gst/rtsp-server/rtsp-client.c:
7511 client: cleanup error paths
7513 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7515 * gst/rtsp-server/rtsp-media.c:
7518 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7520 * examples/test-video.c:
7521 test: enable SRTP only on RTSPS
7522 We only want to enable SRTP when doing rtsp over TLS so that we can
7523 exchange the keys in a secure way.
7525 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7527 * examples/test-video.c:
7528 test: print an error on failure
7530 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7533 * examples/test-video.c:
7534 * gst/rtsp-server/rtsp-sdp.c:
7535 * gst/rtsp-server/rtsp-stream.c:
7536 * tests/check/Makefile.am:
7537 stream: add SRTP support
7538 Install srtp encoder and decoder elements in rtpbin
7541 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7543 * tests/check/Makefile.am:
7544 * tests/check/gst/sessionpool.c:
7545 tests: Add unit tests for sessionpool
7546 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7548 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7550 * tests/check/gst/threadpool.c:
7551 tests: Improve code coverage of rtsp-threadpool tests
7552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7554 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7556 * tests/check/gst/sessionmedia.c:
7557 tests: Improve code coverage for rtsp-session-media
7558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7560 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7562 gobject-introspection: Add annotations to support language bindings
7563 In addition a few cosmetic changes:
7564 * Adjust the order of arguments
7565 * Fix typo: occured -> occurred
7566 * Fix indentation after Return:-clauses
7567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7569 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7571 * gst/rtsp-server/rtsp-stream.c:
7572 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7575 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7577 * gst/rtsp-server/rtsp-stream.c:
7578 stream: take caps after the session manager
7579 Take the caps for the SDP after they leave the rtpbin so that we can
7580 also get the properties added by rtpbin elements.
7582 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7584 * gst/rtsp-server/rtsp-stream.c:
7585 stream: release lock while pushing out packets
7586 Keep a cache of the transports and use this to iterate the transport
7587 while pushing packets. This allows us to release the lock early.
7588 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7590 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7592 * gst/rtsp-server/rtsp-client.c:
7593 * gst/rtsp-server/rtsp-client.h:
7594 rtsp-client: vmethod for modifying tunnel GET response
7595 Add a vmethod tunnel_http_response where the response to the HTTP GET
7596 for tunneled connections can be modified.
7597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7599 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7601 * gst/rtsp-server/rtsp-sdp.c:
7602 sdp: make 1 media line per profile
7603 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7604 line in the SDP for each profile. The client is then supposed to pick
7605 one of the profiles in the SETUP request. Because the m= lines have the
7606 same pt, the client also knows that only 1 option is possible.
7608 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7610 * gst/rtsp-server/rtsp-media-factory.c:
7611 * gst/rtsp-server/rtsp-media-factory.h:
7612 * gst/rtsp-server/rtsp-media.c:
7613 factory: add profile property and pass to media and streams
7615 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7617 * examples/test-multicast.c:
7618 * gst/rtsp-server/rtsp-sdp.c:
7619 sdp: pass multicast connection for multicast-only stream
7620 Pass the multicast address of the stream in the connection info in the
7621 SDP so that clients try a multicast connection first.
7622 Only allow multicast connections in the test-multicast example. Also
7623 increase the TTL a little.
7625 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7628 .gitignore: Ignore gcov intermediate files
7629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7631 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7633 * gst/rtsp-server/rtsp-stream.c:
7634 stream: release some locks in error cases
7636 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7638 docs: Enable and fix gtk-doc warnings
7639 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7640 * addresspool/mediafactory: Add missing annotation colon
7641 * stream: Annotate return value
7642 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7644 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7647 Automatic update of common submodule
7648 From fe1672e to bcb1518
7650 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7653 Automatic update of common submodule
7654 From 1a07da9 to fe1672e
7656 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7658 * examples/Makefile.am:
7659 examples: use LDADD for libs instead of LDFLAGS
7661 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7664 configure: make sure releases are in .doap file
7666 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7668 * examples/test-cgroups.c:
7669 examples: test-cgroups: don't put code with side effects into g_assert()
7670 The g_assert() might get compiled out with the right
7671 compiler/preprocessor flags.
7673 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7675 * examples/.gitignore:
7676 examples: add cgroup test binary to .gitignore
7678 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7680 * examples/test-cgroups.c:
7681 examples: fix cgroup test build
7682 Fixes build failure caused by compiler warning:
7683 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7685 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7688 .gitignore: ignore temp files created in the course of 'make check'
7690 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7692 * gst/rtsp-server/rtsp-media.c:
7693 rtsp-media: don't loose frames handling new PLAY request
7694 If client supplied a range check if the range specifies the start point.
7695 If not, then do an accurate seek to the current position. If a start
7696 point was specified do do a key unit seek to make sure the streaming
7697 starts with decodeable frames.
7698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7700 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7702 * gst/rtsp-server/rtsp-media.c:
7703 Revert "media: only flush when setting a new start position"
7704 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7705 We need to do the flush in all cases, demuxer block currently for
7708 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7710 * gst/rtsp-server/rtsp-media.c:
7711 media: only flush when setting a new start position
7712 Only flush the pipeline when we change the start position with
7714 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7716 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7718 * gst/rtsp-server/rtsp-stream.c:
7719 stream: set ttl-mc before adding the socket
7720 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7721 never be set on socket.
7722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7724 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7726 * gst/rtsp-server/rtsp-media.c:
7727 media: stop thread if media is already prepared
7728 in gst_rtsp_media_prepare() the thread is not used if media is already
7729 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7731 https://bugzilla.gnome.org/show_bug.cgi?id=724182
7733 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
7736 build: Ship gst-rtsp-server.doap file
7738 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
7740 * tests/check/gst/rtspserver.c:
7741 tests: Fix another compiler warning with gcc
7743 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
7745 * gst/rtsp-server/rtsp-client.c:
7746 * gst/rtsp-server/rtsp-mount-points.c:
7747 * gst/rtsp-server/rtsp-stream.c:
7748 * tests/check/gst/client.c:
7749 rtsp-server: Fix lots of compiler warnings with clang
7751 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
7754 * gst-rtsp-server.doap:
7755 * tests/Makefile.am:
7756 configure: Synchronise with the configure scripts of the other modules
7758 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
7761 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
7763 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
7765 * gst/rtsp-server/rtsp-media.c:
7766 * gst/rtsp-server/rtsp-stream.c:
7767 Revert "rtsp-server: support build against last stable release"
7768 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
7769 Let us require 1.2.3 now, which is going to be released in a few
7772 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
7774 * gst/rtsp-server/rtsp-session-media.c:
7775 * gst/rtsp-server/rtsp-stream-transport.c:
7776 session: improve RTP-Info
7777 Ignore streams that can't generate RTP-Info instead of failing.
7778 Don't return the empty string when all streams are unconfigured but
7779 return NULL so that we don't generate and empty RTP-Info header.
7780 Improve docs a little.
7782 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
7784 * gst/rtsp-server/rtsp-session-media.c:
7785 Don't free rtpinfo GString when it is NULL
7786 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7788 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
7790 * gst/rtsp-server/rtsp-media.c:
7791 media: only set keyframe flag when modifying start
7792 Only set the keyframe flag when we modify the start position. The
7793 keyframe flag should probably be ignored when no change is requested but
7794 until we can claim this is all documented properly and all demuxer
7795 implement this, avoid setting the flag.
7796 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
7798 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
7800 * gst/rtsp-server/rtsp-thread-pool.c:
7801 thread-pool: Unref source after mainloop has quit to avoid races in GLib
7802 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
7804 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
7806 * gst/rtsp-server/rtsp-stream.c:
7807 stream: handle NULL seqnum and rtptime arguments
7809 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
7811 * gst/rtsp-server/rtsp-thread-pool.c:
7812 * tests/check/gst/threadpool.c:
7813 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
7814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
7816 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
7818 * gst/rtsp-server/rtsp-stream.c:
7819 stream: add fallback for missing stats property
7820 Use a fallback when the payloader does not have a stats property
7821 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7823 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
7826 Automatic update of common submodule
7827 From f7bc1c3 to 1a07da9
7829 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
7831 * gst/rtsp-server/rtsp-stream.c:
7832 stream: don't leak stats structure
7833 Don't leak the stats structure and deal with NULL stats.
7835 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
7837 * gst/rtsp-server/rtsp-stream.c:
7838 stream: Get rtpinfo properties atomically from payloader
7839 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
7841 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
7843 * gst/rtsp-server/rtsp-media.c:
7844 media: refactor state change functions and signals
7845 Make functions to set the target state and the pipeline state and emit
7846 the signals from those functions.
7848 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
7850 * gst/rtsp-server/rtsp-media.c:
7851 * gst/rtsp-server/rtsp-media.h:
7852 media: add signal to notify of pending state changes
7854 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
7856 * gst/rtsp-server/rtsp-media.c:
7857 * gst/rtsp-server/rtsp-stream.c:
7858 rtsp-server: support build against last stable release
7859 Until 1.2.3 is out with the new get_type function and we
7862 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
7864 * gst/rtsp-server/rtsp-stream.c:
7865 stream: fix compilation
7867 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
7869 * gst/rtsp-server/rtsp-media.c:
7870 * gst/rtsp-server/rtsp-media.h:
7871 * gst/rtsp-server/rtsp-stream.c:
7872 * gst/rtsp-server/rtsp-stream.h:
7873 stream: add property to configure profiles
7875 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
7877 * gst/rtsp-server/rtsp-client.c:
7878 client: let stream check supported transport
7879 Delegate the check if a transport is allowed to the stream.
7880 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
7882 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
7884 * gst/rtsp-server/rtsp-stream.c:
7885 * gst/rtsp-server/rtsp-stream.h:
7886 stream: add method to check supported transport
7887 Add a method to check if a transport is supported
7889 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
7892 configure.ac: Only check for gstreamer-check, not check
7893 We include check in gstreamer-check since quite some time now.
7895 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
7897 * gst/rtsp-server/rtsp-session-media.c:
7898 * gst/rtsp-server/rtsp-stream-transport.c:
7899 * gst/rtsp-server/rtsp-stream.c:
7900 * gst/rtsp-server/rtsp-stream.h:
7901 stream: return clock-rate from get_rtpinfo
7902 And use it to correct the rtptime to the requested start-time.
7903 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
7905 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
7907 * gst/rtsp-server/rtsp-session-media.c:
7908 * gst/rtsp-server/rtsp-stream-transport.c:
7909 * gst/rtsp-server/rtsp-stream-transport.h:
7910 session-media: calculate start-time
7912 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
7914 * gst/rtsp-server/rtsp-stream-transport.c:
7915 * gst/rtsp-server/rtsp-stream.c:
7916 * gst/rtsp-server/rtsp-stream.h:
7917 stream: also return the running-time
7918 Return the running-time in the rtpinfo as well.
7920 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
7922 * gst/rtsp-server/rtsp-client.c:
7923 * gst/rtsp-server/rtsp-session-media.c:
7924 * gst/rtsp-server/rtsp-session-media.h:
7925 * gst/rtsp-server/rtsp-stream-transport.c:
7926 * gst/rtsp-server/rtsp-stream-transport.h:
7927 session-media: let the session-media make the RTPInfo
7928 Add method to create the RTPInfo for a stream-transport.
7929 Add method to create the RTPInfo for all stream-transports in a
7931 Use the session-media RTPInfo code in client. This allows us to refactor
7932 another method to link the TCP callbacks.
7934 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7936 mount-points: sort sequence before g_sequence_lookup
7937 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
7938 sort sequence if dirty, otherwise lookup will fail.
7939 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
7941 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
7944 configure: rename package from gst-rtsp to gst-rtsp-server
7945 To match git module name and avoid confusion with the
7946 rtsp lib in gst-plugins-base and rtsp plugin in -good.
7948 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
7951 configure: bump core/base/good requirement to 1.2.0
7952 Bump to released stable version and make implicit
7953 requirements explicit.
7955 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
7960 Fix broken gettext setup which is not used anyway
7962 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
7965 Automatic update of common submodule
7966 From dbedaa0 to d48bed3
7968 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
7970 * gst/rtsp-server/rtsp-client.c:
7971 * gst/rtsp-server/rtsp-media.c:
7972 * gst/rtsp-server/rtsp-media.h:
7973 media: add setup_sdp vmethod
7974 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
7975 gst_rtsp_media_setup_sdp.
7976 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
7978 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
7980 * gst/rtsp-server/rtsp-stream.c:
7981 rtsp-stream: Check return value of sscanf
7982 streamid is only valid if sscanf matched something.
7984 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
7986 * gst/rtsp-server/rtsp-client.c:
7987 rtsp-client: Fix iteration
7988 Wouldn't even enter the code block otherwise (i++ was used as the check
7989 and not the postfix).
7991 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
7993 * gst/rtsp-server/rtsp-client.c:
7994 * gst/rtsp-server/rtsp-client.h:
7995 client: add vmethod to configure media and streams
7996 Implement a vmethod that can be used to configure the media and the
7997 streams based on the current context. Handle the blocksize handling in
7998 the default handler.
7999 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
8001 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8004 Make git ignore more unit test binaries
8006 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8008 * gst/rtsp-server/rtsp-address-pool.h:
8009 * gst/rtsp-server/rtsp-auth.h:
8010 * gst/rtsp-server/rtsp-client.h:
8011 * gst/rtsp-server/rtsp-context.h:
8012 * gst/rtsp-server/rtsp-media-factory-uri.h:
8013 * gst/rtsp-server/rtsp-media-factory.h:
8014 * gst/rtsp-server/rtsp-media.h:
8015 * gst/rtsp-server/rtsp-mount-points.h:
8016 * gst/rtsp-server/rtsp-server.h:
8017 * gst/rtsp-server/rtsp-session-media.h:
8018 * gst/rtsp-server/rtsp-session-pool.h:
8019 * gst/rtsp-server/rtsp-session.h:
8020 * gst/rtsp-server/rtsp-stream-transport.h:
8021 * gst/rtsp-server/rtsp-stream.h:
8022 * gst/rtsp-server/rtsp-thread-pool.h:
8023 * gst/rtsp-server/rtsp-token.h:
8024 rtsp-server: add padding to many public structures
8025 Not mini objects though, since they are not subclassable
8026 anyway, nor kept on the stack or inlined in a structure.
8028 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8030 media: add new create_rtpbin vmethod
8031 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8032 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8034 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8036 * tests/check/gst/media.c:
8037 tests: fix memory leak, free test's thread pool
8038 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8040 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8042 * gst/rtsp-server/rtsp-stream-transport.c:
8043 stream-transport: free url in finalize
8045 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8047 * gst/rtsp-server/rtsp-media.c:
8048 media: also do state change in suspended state
8050 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8052 * gst/rtsp-server/rtsp-client.c:
8053 * gst/rtsp-server/rtsp-media.c:
8054 media: also handle prepare and range in suspended state
8055 When we are suspended, we are already prepared.
8056 We can get the range in the suspended state.
8058 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8060 * tests/check/Makefile.am:
8061 * tests/check/gst/sessionmedia.c:
8062 check: add test for uri in setup
8063 Added unit tests for the new functionality in GstRTSPStreamTransport.
8064 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8066 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8068 * gst/rtsp-server/rtsp-client.c:
8069 client: store setup uri and use in PLAY response
8070 Store the uri used when doing the setup and use that in the PLAY
8072 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8074 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8076 * gst/rtsp-server/rtsp-stream-transport.c:
8077 * gst/rtsp-server/rtsp-stream-transport.h:
8078 stream-transport: add method to get/set url
8080 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8082 * gst/rtsp-server/rtsp-client.c:
8083 client: suspend after SDP and unsuspend before PLAYING
8084 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8087 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8089 * gst/rtsp-server/rtsp-media-factory.c:
8090 * gst/rtsp-server/rtsp-media-factory.h:
8091 * gst/rtsp-server/rtsp-media.c:
8092 * gst/rtsp-server/rtsp-media.h:
8093 * gst/rtsp-server/rtsp-session-media.c:
8094 * gst/rtsp-server/rtsp-session.c:
8095 * tests/check/gst/media.c:
8096 * tests/check/gst/mediafactory.c:
8097 media: add suspend modes
8098 Add support for different suspend modes. The stream is suspended right after
8099 producing the SDP and after PAUSE. Different suspend modes are available that
8100 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8101 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8102 state and RESET will bring the pipeline to the NULL state.
8103 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8104 this means that the pipeline needs to be prerolled again.
8105 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8106 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8108 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8110 * gst/rtsp-server/rtsp-media.c:
8111 media: start live streams in blocked state
8112 Start live streams in the blocked state and make them preroll using the
8113 messages. This ensure that no data is played by the sink until we explicitly
8114 unblock the stream right before going to PLAYING.
8115 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8117 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8119 * gst/rtsp-server/rtsp-media.c:
8120 media: refactor starting and waiting for preroll
8121 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8122 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8124 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8126 * gst/rtsp-server/rtsp-stream.c:
8127 * gst/rtsp-server/rtsp-stream.h:
8128 stream: add API to block streams
8129 Add an API to block on the streams and make it post a message.
8130 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8131 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8133 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8135 * docs/libs/Makefile.am:
8136 docs: Specify the override file
8137 Even if it's empty (for now) it avoids make distcheck complaining
8139 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8141 * gst/rtsp-server/rtsp-media.c:
8142 media: move default implementations to where they are used
8144 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8146 * gst/rtsp-server/rtsp-media.c:
8147 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8148 We need to take the state_lock when calling this method.
8150 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8152 * gst/rtsp-server/rtsp-media.c:
8153 media: handle add-added on non-bins too
8154 Handle dynamic payloaders that are not bins, as used in the unit-test.
8156 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8158 * gst/rtsp-server/rtsp-media-factory.c:
8159 * gst/rtsp-server/rtsp-media-factory.h:
8160 * gst/rtsp-server/rtsp-media.c:
8161 rtsp-media/-factory: Fix request pad name comments
8162 These must be escaped for gtk-doc to parse the comments without warnings.
8164 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8166 rtsp-media: remove transports if media is in error status
8167 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8168 trying to change to GST_STATE_NULL and media is in error status, we
8169 remove all transports.
8170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8172 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8174 * gst/rtsp-server/rtsp-media.c:
8175 rtsp-media: use element metadata to find payloader
8176 Use the element metadata to find the payloader instead of checking
8178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8180 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8182 rtsp-stream: add getter for payload type
8183 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8184 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8185 element and create the stream with this one instead of the dynpay%d
8187 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8189 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8191 * gst/rtsp-server/rtsp-client.c:
8192 * gst/rtsp-server/rtsp-context.h:
8193 * gst/rtsp-server/rtsp-media.c:
8194 * gst/rtsp-server/rtsp-mount-points.c:
8195 * gst/rtsp-server/rtsp-server.c:
8196 * gst/rtsp-server/rtsp-token.c:
8197 rtsp-*: Refer to NULL as a constant in comments
8199 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8201 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8203 rtsp-*: Fix type name typos in comments
8204 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8205 * rtsp-auth: Refer to part of constant name as text
8206 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8207 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8208 * rtsp-stream: Fix typo when refering to GstBin
8209 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8211 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8214 * docs/libs/gst-rtsp-server-docs.sgml:
8215 * docs/libs/gst-rtsp-server-sections.txt:
8216 docs: Improve documentation
8217 * Include annotation-glossary to quiet gtk-doc
8218 * Rename remaining ClientState -> Context
8219 * Rename object hierarchy file
8220 * Remove stale chapter references
8221 * Add missing function and object references
8222 * Include missing GstRTSPAddressPoolResult
8223 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8225 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.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 * gst/rtsp-server/rtsp-stream.c:
8232 rtsp-server: sprinkle some allow-none annotations for g-i
8234 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8236 * gst/rtsp-server/rtsp-stream.c:
8237 * gst/rtsp-server/rtsp-stream.h:
8238 stream: add method to filter transports
8239 Add a method to safely iterate and collect the stream transports
8240 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8242 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8244 * gst/rtsp-server/rtsp-client.c:
8245 * gst/rtsp-server/rtsp-server.c:
8246 * gst/rtsp-server/rtsp-session-pool.c:
8247 * gst/rtsp-server/rtsp-session.c:
8248 rtsp: allow NULL func in filters
8249 Passing a null function make the filters return a list of
8252 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8254 * gst/rtsp-server/rtsp-address-pool.c:
8255 * tests/check/gst/addresspool.c:
8256 address-pool: fix address increment
8257 Use a guint instead of guint8 to increment the address. It's still not
8258 completely correct because a guint might not be able to hold the complete
8259 address range, but that's an enhacement for later.
8260 Add unit test to test improved behaviour.
8261 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8263 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8265 * gst/rtsp-server/rtsp-client.c:
8266 * tests/check/gst/client.c:
8267 client: allow absolute path in requests
8268 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8270 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8272 * gst/rtsp-server/rtsp-client.c:
8273 * gst/rtsp-server/rtsp-client.h:
8274 client: make make_path_from_uri a vmethod
8276 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8278 * docs/libs/gst-rtsp-server-sections.txt:
8279 * gst/rtsp-server/rtsp-stream.c:
8280 * gst/rtsp-server/rtsp-stream.h:
8281 * tests/check/Makefile.am:
8282 * tests/check/gst/stream.c:
8283 stream: Add functions to get rtp and rtcp sockets
8284 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8286 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8288 * gst/rtsp-server/rtsp-context.c:
8289 * gst/rtsp-server/rtsp-context.h:
8290 context: defing a GType for the context
8291 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8293 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8295 * gst/rtsp-server/Makefile.am:
8296 * gst/rtsp-server/rtsp-auth.c:
8297 * gst/rtsp-server/rtsp-context.c:
8298 * gst/rtsp-server/rtsp-media.c:
8299 * gst/rtsp-server/rtsp-mount-points.c:
8300 * gst/rtsp-server/rtsp-server.h:
8301 * gst/rtsp-server/rtsp-session-media.c:
8302 * gst/rtsp-server/rtsp-session.c:
8303 * gst/rtsp-server/rtsp-stream.c:
8304 Fixed several GIR warnings
8306 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8308 * gst/rtsp-server/rtsp-auth.c:
8311 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8313 * tests/check/Makefile.am:
8314 * tests/check/gst/token.c:
8315 tests: Add unit tests for token
8316 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8318 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8320 * gst/rtsp-server/rtsp-token.c:
8321 token: Validate args for gst_rtsp_token_is_allowed
8322 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8324 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8326 * gst/rtsp-server/rtsp-token.c:
8327 token: Fix bug when creating empty token
8328 We always want to have a valid GstStructure in the token.
8329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8331 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8333 * gst/rtsp-server/rtsp-thread-pool.c:
8334 thread-pool: avoid race in shutdown
8335 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8336 don't actually stop the mainloop ever. Solve this race by adding an idle source
8337 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8338 if quit was called before we started it.
8340 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8342 * tests/check/Makefile.am:
8343 * tests/check/gst/permissions.c:
8344 tests: Add unit tests for permissions
8345 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8347 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8349 * tests/check/gst/mediafactory.c:
8350 tests: Test mediafactory permissions
8351 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8353 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8355 * gst/rtsp-server/rtsp-permissions.c:
8356 permissions: Fix refcounting when adding/removing roles
8357 Previously a role that was removed was unreffed twice, and when
8358 replacing an existing role the replaced role was freed while still being
8359 referenced. Both bugs are now fixed.
8360 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8362 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8364 * tests/check/gst/media.c:
8365 * tests/check/gst/mediafactory.c:
8366 * tests/check/gst/rtspserver.c:
8367 tests: Check gst_rtsp_url_parse return value
8368 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8370 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8373 Automatic update of common submodule
8374 From 865aa20 to dbedaa0
8376 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8378 * gst/rtsp-server/rtsp-server.c:
8379 rtsp-server: Fix socket leak
8380 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8382 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8384 * gst/rtsp-server/rtsp-session-pool.c:
8385 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8386 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8388 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8390 * examples/.gitignore:
8391 * examples/test-video.c:
8392 examples: fix compilation when WITH_AUTH is defined
8393 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8395 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8398 gitignore: Add new test binary
8400 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8402 * tests/check/Makefile.am:
8403 * tests/check/gst/threadpool.c:
8404 thread-pool: Add unit test for the thread pools
8405 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8407 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8409 * gst/rtsp-server/rtsp-thread-pool.c:
8410 thread-pool: Fix thread leak when reusing threads
8411 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8413 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8415 * gst/rtsp-server/rtsp-server.c:
8416 * tests/check/gst/rtspserver.c:
8417 tests: fixed racy behavior in rtspserver tests
8418 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8420 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8422 * tests/check/gst/addresspool.c:
8423 tests: Improve address pool unit tests
8424 Add a range with mixed IPV4 and IPV6 addresses to pool.
8425 Get an IPV4 address from an IPV6-only pool.
8426 Get an IPV6 address from an IPV4-only pool.
8427 Reserve a IPV6 address from an IPV4-only pool.
8428 Check for unicast addresses in multicast-only pool.
8429 Check for unicast addresses in uni-/multicast-mixed pool.
8430 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8432 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8434 * gst/rtsp-server/rtsp-client.c:
8435 client: append query string in PAUSE/PLAY/TEARDOWN as well
8437 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8439 * gst/rtsp-server/rtsp-client.c:
8440 client: Add query to control path
8441 If the SETUP url contains a query it must be appended to the control
8442 path so that it matches any already created stream in the media. The
8443 query will also be appended to the session media path.
8445 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8447 * gst/rtsp-server/rtsp-media.c:
8448 rtsp-media: remove old line
8450 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8452 * gst/rtsp-server/rtsp-stream.c:
8453 stream: Correct control comparison
8454 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8456 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8458 * gst/rtsp-server/rtsp-media.c:
8459 media: Check dynamically if the pipeline supports seeking
8460 We should not depend on whether or not the pipeline state change
8461 returned NO_PREROLL or not. A media could dynamically change its
8462 element and switch from seekable to non seekable so it's best to test
8463 the seekable nature of the pipeline dynamically when we try to do a seek.
8465 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-media.c:
8468 media: Return FALSE if seeking is not supported
8470 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8472 * gst/rtsp-server/rtsp-media.c:
8473 rtsp-media: don't seek accurate by default
8474 Accurate seeking is perhaps a little overkill in the most common situation and
8475 causes some formats (mp3) over slow media to seek extremely slowly.
8477 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8479 * tests/check/gst/rtspserver.c:
8480 tests: fix unit test
8481 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8483 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8485 * gst/rtsp-server/rtsp-client.c:
8486 client: Reply 400 if media cannot be constructed
8487 Reply 400 Bad Request instead of 503 Service Unavailable if media
8488 cannot be constructed in SETUP.
8489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8491 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8493 * gst/rtsp-server/rtsp-client.c:
8494 client: Send setup reply once only
8495 If find_media() failed in handle_setup_request() two replies was sent.
8496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8498 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8501 Automatic update of common submodule
8502 From 6b03ba7 to 865aa20
8504 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8506 * gst/rtsp-server/rtsp-server.c:
8507 server: Emit client-connected signal earlier
8508 Emit client-connected before the client ref is given to a GSource,
8509 otherwise client-connected can be emitted after the client object has
8512 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8514 * gst/rtsp-server/rtsp-address-pool.c:
8515 * gst/rtsp-server/rtsp-address-pool.h:
8516 * gst/rtsp-server/rtsp-stream.c:
8517 * tests/check/gst/addresspool.c:
8518 addresspool: return reason of failure
8519 Let gst_rtsp_address_pool_reserve_address() return the reason why
8520 the address could not be reserved.
8521 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8523 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8526 autogen.sh: Sync behaviour with other GStreamer modules
8527 Allows building from outside of tree amongst other things
8529 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8532 Automatic update of common submodule
8533 From b613661 to 6b03ba7
8535 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8538 Automatic update of common submodule
8539 From 74a6857 to b613661
8541 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8544 Automatic update of common submodule
8545 From 01a7a46 to 74a6857
8547 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8549 * gst/rtsp-server/rtsp-client.c:
8550 client: Do not read beyond end of path string
8551 If the setup was done without a control url, make sure we don't try to read the
8552 non-existing control string and crash.
8554 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8556 * gst/rtsp-server/rtsp-client.c:
8557 client: Fix RTPInfo header
8558 Refactor the method to make the content_base.
8559 Use the content-base and the control url to construct the RTPInfo
8562 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8564 * gst/rtsp-server/rtsp-client.c:
8565 client: map url to path only in describe
8566 Only map the request url to a path in the DESCRIBE method. The SDP then
8567 contains the base and control urls that should be used to SETUP/PAUSE/
8568 PLAY/TEARDOWN the media.
8570 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8572 * gst/rtsp-server/rtsp-client.c:
8573 Revert "client: map URL to path in requests"
8574 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8575 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8576 contains the base and control urls which are used in the SETUP, PLAY,
8577 PAUSE and TEARDOWN requests.
8579 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8581 * gst/rtsp-server/rtsp-client.c:
8582 client: map URL to path in requests
8584 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/rtsp-client.c:
8587 * gst/rtsp-server/rtsp-mount-points.c:
8588 * gst/rtsp-server/rtsp-mount-points.h:
8589 mount-points: make vmethod to make path from uri
8590 Make a vmethod to transform an url into a path. The path is then used to lookup
8591 the factory. This makes it possible to also use other bits of the url, such as
8592 the query parameters, to locate the factory.
8594 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8596 * gst/rtsp-server/rtsp-thread-pool.c:
8597 * gst/rtsp-server/rtsp-thread-pool.h:
8598 thread-pool: Add cleanup to wait for the threadpool to finish
8599 Also fix race condition if two threads are asking for the first
8600 thread from the thread pool at once. This would case two internal
8601 GThreadPools to be created.
8602 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8604 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8606 * gst/rtsp-server/rtsp-client.c:
8607 * tests/check/gst/client.c:
8608 client: free threadpool
8609 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8611 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8613 * tests/check/gst/mountpoints.c:
8614 mountpoints tests: unref matched factories
8615 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8617 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8619 * tests/check/gst/media.c:
8620 media tests: unref thread pool and caps
8621 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8623 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8625 * gst/rtsp-server/rtsp-auth.c:
8626 * gst/rtsp-server/rtsp-media-factory.c:
8627 * gst/rtsp-server/rtsp-media.c:
8628 auth, media, media-factory: unref permissions
8629 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8631 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8633 * examples/Makefile.am:
8634 Makefile: add rule for appsrc example
8636 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8638 * examples/test-appsrc.c:
8639 tests: add appsrc example
8640 Add an example on how to use appsrc to feed the server pipeline with data.
8642 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8644 * gst/rtsp-server/rtsp-client.c:
8645 rtsp-client: remove query part from content-base string
8646 Make sure that after the control url has been resolved, it's
8647 not a part of the query-string.
8648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8650 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8652 * gst/rtsp-server/rtsp-client.c:
8653 client: don't check url in response
8654 There is no url or method in the response to check
8656 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8658 * gst/rtsp-server/rtsp-client.c:
8659 * gst/rtsp-server/rtsp-client.h:
8660 Add handle-response signal for when we receive a GET_PARAMETER response
8662 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8664 * gst/rtsp-server/rtsp-server.c:
8665 Fix gst_rtsp_server_client_filter, using wrong variable type
8667 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8669 * gst/rtsp-server/rtsp-media-factory-uri.c:
8670 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8671 For AAC we need to check for framed=true instead of parsed=true.
8672 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8674 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8676 * gst/rtsp-server/rtsp-stream.c:
8677 stream: optimize pipeline for protocols
8678 When TCP is not an allowed protocol for the stream, avoid creating the
8679 appsrc/appsink/queue and tee elements.
8681 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8683 * gst/rtsp-server/rtsp-media.c:
8684 media: set protocols on streams
8686 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8688 * gst/rtsp-server/rtsp-client.c:
8689 client: use protocols supported by stream
8691 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8693 * gst/rtsp-server/rtsp-media-factory.c:
8694 * gst/rtsp-server/rtsp-media.c:
8695 * gst/rtsp-server/rtsp-stream.c:
8696 media-factory: allow all protocols
8698 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8700 * gst/rtsp-server/rtsp-media.c:
8701 media: configure protocols in new streams
8703 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8705 * gst/rtsp-server/rtsp-stream.c:
8706 * gst/rtsp-server/rtsp-stream.h:
8707 stream: add protocols property
8709 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8711 * gst/rtsp-server/rtsp-media.c:
8712 rtsp-media: send state in "new-state" signal
8713 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8715 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8718 build: add subdir-objects to AM_INIT_AUTOMAKE
8719 Fixes warnings with automake 1.14
8720 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8722 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8724 * docs/libs/gst-rtsp-server-sections.txt:
8725 * gst/rtsp-server/rtsp-client.c:
8726 * gst/rtsp-server/rtsp-server.c:
8727 * gst/rtsp-server/rtsp-server.h:
8728 server: add method to iterate clients of server
8730 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8732 * gst/rtsp-server/rtsp-media.c:
8733 * gst/rtsp-server/rtsp-media.h:
8734 Add vmethod for rtsp-media subclass to access rtpbin
8736 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8738 * gst/rtsp-server/rtsp-client.h:
8739 small documentation fix
8741 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-client.c:
8744 Do not take range header if range is invalid
8746 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8748 * docs/libs/gst-rtsp-server-sections.txt:
8749 * gst/rtsp-server/rtsp-media.c:
8750 media: add docs for new method
8752 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8754 * gst/rtsp-server/rtsp-media.c:
8755 * gst/rtsp-server/rtsp-media.h:
8756 Add API to rtsp-media set the pipeline's state
8758 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8760 * gst/rtsp-server/rtsp-media.c:
8761 Update current position/duration when gst_rtsp_media_get_range_string is called
8763 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8765 * examples/test-cgroups.c:
8766 tests: add some more docs
8768 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8770 * examples/test-cgroups.c:
8771 * gst/rtsp-server/Makefile.am:
8772 * gst/rtsp-server/rtsp-auth.c:
8773 * gst/rtsp-server/rtsp-auth.h:
8774 * gst/rtsp-server/rtsp-client.c:
8775 * gst/rtsp-server/rtsp-client.h:
8776 * gst/rtsp-server/rtsp-context.c:
8777 * gst/rtsp-server/rtsp-context.h:
8778 * gst/rtsp-server/rtsp-params.c:
8779 * gst/rtsp-server/rtsp-params.h:
8780 * gst/rtsp-server/rtsp-server.c:
8781 * gst/rtsp-server/rtsp-thread-pool.c:
8782 * gst/rtsp-server/rtsp-thread-pool.h:
8783 * tests/check/gst/client.c:
8784 ClientState -> Context
8785 Rename the clientstate to context and put the code in a separate file.
8787 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8789 * examples/test-auth.c:
8790 * gst/rtsp-server/rtsp-auth.c:
8791 * gst/rtsp-server/rtsp-auth.h:
8792 auth: add support for default token
8793 The default token is used when the user is not authenticated and can be used to
8794 give minimal permissions.
8796 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8798 * examples/test-auth.c:
8799 * gst/rtsp-server/rtsp-auth.c:
8800 auth: use defines when possible
8802 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8804 * gst/rtsp-server/rtsp-address-pool.c:
8805 address-pool: improve docs
8807 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8809 * gst/rtsp-server/rtsp-permissions.c:
8810 permissions: add the role to the copy
8812 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
8814 * gst/rtsp-server/rtsp-permissions.c:
8815 permissions: Also copy the roles
8817 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
8819 * gst/rtsp-server/rtsp-permissions.c:
8820 permissions: Make it build
8822 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8824 * gst/rtsp-server/rtsp-address-pool.h:
8827 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8829 * docs/libs/gst-rtsp-server-sections.txt:
8830 * gst/rtsp-server/rtsp-auth.c:
8831 * gst/rtsp-server/rtsp-auth.h:
8832 * gst/rtsp-server/rtsp-media.c:
8833 * gst/rtsp-server/rtsp-session-media.c:
8834 * gst/rtsp-server/rtsp-stream-transport.c:
8835 * gst/rtsp-server/rtsp-stream-transport.h:
8836 * gst/rtsp-server/rtsp-stream.c:
8837 * tests/check/gst/client.c:
8840 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8842 * docs/libs/gst-rtsp-server-sections.txt:
8843 * gst/rtsp-server/rtsp-address-pool.c:
8844 * gst/rtsp-server/rtsp-address-pool.h:
8845 * tests/check/gst/addresspool.c:
8846 * tests/check/gst/rtspserver.c:
8847 address-pool: cleanups
8848 Remove redundant method, improve docs.
8850 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8852 * docs/libs/gst-rtsp-server-sections.txt:
8853 * gst/rtsp-server/rtsp-auth.h:
8854 * gst/rtsp-server/rtsp-permissions.c:
8855 * gst/rtsp-server/rtsp-permissions.h:
8856 * gst/rtsp-server/rtsp-token.c:
8859 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8861 * gst/rtsp-server/rtsp-permissions.c:
8862 permissions: implement _remove_role
8864 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8866 * gst/rtsp-server/rtsp-permissions.c:
8867 permissions: update docs
8869 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8871 * tests/check/gst/client.c:
8872 tests: simplify tests
8873 Client settings are now disabled by default so we don't need an auth
8874 module to disable them.
8876 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8878 * gst/rtsp-server/rtsp-auth.c:
8879 auth: add default authorizations
8880 When no auth module is specified, use our table of defaults to look up the
8881 default value of the check instead of always allowing everything. This was
8882 we can disallow client settings by default.
8884 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8887 README: update readme
8889 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8891 * gst/rtsp-server/rtsp-thread-pool.c:
8892 * gst/rtsp-server/rtsp-thread-pool.h:
8893 thread-pool: add more docs
8895 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8897 * gst/rtsp-server/rtsp-thread-pool.c:
8898 * gst/rtsp-server/rtsp-thread-pool.h:
8899 thread-pool: fix race in thread reuse
8900 If we try to reuse a thread right after we made it stop, we end up using a
8901 stopped thread. Catch this case and only reuse threads that are not stopping.
8903 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8905 * gst/rtsp-server/rtsp-server.c:
8906 server: add small debug
8908 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8910 * tests/check/gst/client.c:
8912 Add some permissions to media so we can use the auth and enable
8915 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8917 * gst/rtsp-server/rtsp-client.c:
8918 client: support pushed context in handle_request
8919 If we already have a pushed state, reuse it and add our own things. This makes
8920 it easier to write tests.
8922 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8924 * gst/rtsp-server/rtsp-auth.c:
8925 auth: don't auth on methods
8926 Don't authorize on methods anymore but on the resources that we
8927 try to access, this is more flexible.
8928 Move the authorization checks to where they are needed and let the
8929 check return the response on error.
8931 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8933 * gst/rtsp-server/rtsp-mount-points.c:
8934 mount-points: add some debug
8936 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8938 * tests/check/gst/client.c:
8939 tests: almost fix test
8941 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8943 * gst/rtsp-server/rtsp-auth.c:
8944 * gst/rtsp-server/rtsp-auth.h:
8945 * gst/rtsp-server/rtsp-client.c:
8946 * gst/rtsp-server/rtsp-client.h:
8947 * gst/rtsp-server/rtsp-server.c:
8948 * gst/rtsp-server/rtsp-server.h:
8949 auth: let the auth module check client_settings
8950 Let the auth module decide if client settings are allowed for the
8953 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8955 * gst/rtsp-server/rtsp-token.c:
8956 * gst/rtsp-server/rtsp-token.h:
8957 token: add method to check boolean permission
8959 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8961 * examples/test-auth.c:
8962 * examples/test-cgroups.c:
8963 * gst/rtsp-server/rtsp-token.c:
8964 * gst/rtsp-server/rtsp-token.h:
8965 token: simplify token constructor
8966 Use variable arguments to make easier API.
8968 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * examples/test-auth.c:
8971 * examples/test-cgroups.c:
8972 * gst/rtsp-server/rtsp-media-factory.c:
8973 * gst/rtsp-server/rtsp-media-factory.h:
8974 media-factory: add convenience API for factory
8976 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8978 * examples/test-auth.c:
8979 * examples/test-cgroups.c:
8980 * gst/rtsp-server/rtsp-permissions.c:
8981 * gst/rtsp-server/rtsp-permissions.h:
8982 permissions: simplify API a little
8983 Avoid passing GstStructure in the add_role method, use varargs instead
8984 to construct the structure behind the scenes. We can then also use the
8985 structure name as the role and simplify some more logic.
8987 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8989 * gst/rtsp-server/rtsp-auth.c:
8992 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8994 * gst/rtsp-server/rtsp-auth.c:
8995 * gst/rtsp-server/rtsp-auth.h:
8996 * gst/rtsp-server/rtsp-client.c:
8997 auth: handle unauthorized response
8998 Move handling of the unauthorized response to the auth module, it can add
8999 the appropriate headers to request authorization for the required method
9000 much better than the client.
9002 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9004 * gst/rtsp-server/rtsp-client.c:
9005 * gst/rtsp-server/rtsp-client.h:
9006 client: allow for sending any message, not only requests
9007 Change the _send_request() method to _send_message() so that we
9008 can both send requests and replies.
9010 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9012 * docs/libs/gst-rtsp-server-sections.txt:
9013 * gst/rtsp-server/rtsp-server.h:
9016 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9018 * examples/test-video.c:
9019 * gst/rtsp-server/rtsp-auth.c:
9020 * gst/rtsp-server/rtsp-auth.h:
9021 * gst/rtsp-server/rtsp-server.c:
9022 * gst/rtsp-server/rtsp-server.h:
9023 auth: move TLS handling to auth module
9024 Remove the TLS settings on the server and move it to the auth module because
9025 that is where security related bits go.
9027 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9029 * gst/rtsp-server/rtsp-client.c:
9030 * gst/rtsp-server/rtsp-client.h:
9031 client: add state push/pop
9033 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9035 * gst/rtsp-server/rtsp-client.c:
9036 * gst/rtsp-server/rtsp-client.h:
9037 client: add connection to state
9039 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9041 * gst/rtsp-server/rtsp-mount-points.c:
9042 mount-points: fix debug
9044 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * tests/check/gst/media.c:
9047 tests: fix media test
9049 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9051 * gst/rtsp-server/rtsp-thread-pool.c:
9052 thread-pool: we don't require a state
9054 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9056 * gst/rtsp-server/rtsp-server.c:
9057 server: let context ref the server
9058 So that we don't risk losing the server object early anc crash.
9060 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9062 * tests/check/gst/client.c:
9063 tests: fix client test
9065 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9068 * docs/libs/gst-rtsp-server-docs.sgml:
9069 * docs/libs/gst-rtsp-server-sections.txt:
9070 * gst/rtsp-server/rtsp-address-pool.c:
9071 * gst/rtsp-server/rtsp-auth.c:
9072 * gst/rtsp-server/rtsp-client.c:
9073 * gst/rtsp-server/rtsp-client.h:
9074 * gst/rtsp-server/rtsp-media-factory-uri.c:
9075 * gst/rtsp-server/rtsp-media-factory.c:
9076 * gst/rtsp-server/rtsp-media-factory.h:
9077 * gst/rtsp-server/rtsp-media.c:
9078 * gst/rtsp-server/rtsp-mount-points.c:
9079 * gst/rtsp-server/rtsp-params.c:
9080 * gst/rtsp-server/rtsp-permissions.c:
9081 * gst/rtsp-server/rtsp-sdp.c:
9082 * gst/rtsp-server/rtsp-server.c:
9083 * gst/rtsp-server/rtsp-server.h:
9084 * gst/rtsp-server/rtsp-session-media.c:
9085 * gst/rtsp-server/rtsp-session-pool.c:
9086 * gst/rtsp-server/rtsp-session.c:
9087 * gst/rtsp-server/rtsp-stream-transport.c:
9088 * gst/rtsp-server/rtsp-stream.c:
9089 * gst/rtsp-server/rtsp-thread-pool.c:
9090 * gst/rtsp-server/rtsp-token.c:
9093 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9095 * gst/rtsp-server/rtsp-session-pool.c:
9096 * gst/rtsp-server/rtsp-session-pool.h:
9097 session-pool: make vmethod to create a session
9098 Make a vmethod to create a sessions so that subclasses can create
9099 custom session objects
9101 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9103 * gst/rtsp-server/rtsp-auth.c:
9104 * gst/rtsp-server/rtsp-media-factory.h:
9105 * gst/rtsp-server/rtsp-media.h:
9106 * gst/rtsp-server/rtsp-mount-points.h:
9107 * gst/rtsp-server/rtsp-session-pool.h:
9108 * gst/rtsp-server/rtsp-stream.h:
9111 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9113 * docs/libs/gst-rtsp-server-docs.sgml:
9114 * docs/libs/gst-rtsp-server-sections.txt:
9115 * gst/rtsp-server/rtsp-address-pool.c:
9116 * gst/rtsp-server/rtsp-address-pool.h:
9117 * gst/rtsp-server/rtsp-auth.c:
9118 * gst/rtsp-server/rtsp-client.h:
9119 * gst/rtsp-server/rtsp-media-factory.h:
9120 * gst/rtsp-server/rtsp-media.c:
9121 * gst/rtsp-server/rtsp-media.h:
9122 * gst/rtsp-server/rtsp-permissions.c:
9123 * gst/rtsp-server/rtsp-permissions.h:
9124 * gst/rtsp-server/rtsp-server.h:
9125 * gst/rtsp-server/rtsp-session-media.c:
9126 * gst/rtsp-server/rtsp-session-media.h:
9127 * gst/rtsp-server/rtsp-session-pool.h:
9128 * gst/rtsp-server/rtsp-session.h:
9129 * gst/rtsp-server/rtsp-stream-transport.h:
9130 * gst/rtsp-server/rtsp-stream.c:
9131 * gst/rtsp-server/rtsp-thread-pool.h:
9134 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9137 * examples/Makefile.am:
9138 configure: compile cgroup example conditionally
9139 Only compile the cgroup example when we have libcgroup
9141 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9144 * examples/Makefile.am:
9145 * examples/test-cgroups.c:
9146 examples: add cgroups example
9148 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9150 * tests/check/gst/rtspserver.c:
9151 tests: fix compilation
9153 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9155 * gst/rtsp-server/rtsp-thread-pool.c:
9156 thread-pool: fix vmethod invocation
9158 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9160 * gst/rtsp-server/rtsp-thread-pool.c:
9161 * gst/rtsp-server/rtsp-thread-pool.h:
9162 thread-pool: store thread type in thread
9164 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9166 * gst/rtsp-server/rtsp-client.c:
9167 client: pass thread from pool to media _prepare
9168 Get a thread from the configured threadpool and pass it to the prepare method of
9171 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9173 * gst/rtsp-server/rtsp-media.c:
9174 * gst/rtsp-server/rtsp-media.h:
9175 media: Accept a thread in _prepare
9176 Remove out own threadpool handling and use the provided thread and
9177 maincontext for the bus messages and the state changes.
9179 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9181 * gst/rtsp-server/rtsp-server.c:
9182 server: configure client thread pool
9184 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9186 * gst/rtsp-server/rtsp-client.c:
9187 * gst/rtsp-server/rtsp-client.h:
9188 client: add method to configure thread pool
9190 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9192 * gst/rtsp-server/rtsp-client.h:
9193 * gst/rtsp-server/rtsp-server.c:
9194 * gst/rtsp-server/rtsp-server.h:
9195 server: use thread pool
9196 Use the thread pool instead of doing our own thing.
9198 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9200 * gst/rtsp-server/Makefile.am:
9201 * gst/rtsp-server/rtsp-thread-pool.c:
9202 * gst/rtsp-server/rtsp-thread-pool.h:
9203 thread-pool: add object to manage threads
9204 Add an object to manage the client and media threads.
9206 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9208 * gst/rtsp-server/rtsp-auth.c:
9209 auth: debug authorization check
9211 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9213 * gst/rtsp-server/rtsp-media.c:
9214 media: start media pipeline in context
9215 Start the media pipeline in the provided context (or our default one
9216 when NULL). This makes sure that we run the bus thread in this context and that
9217 all media threads are children of this context.
9219 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9221 * gst/rtsp-server/rtsp-media-factory.c:
9222 factory: pass permissions to media by default
9224 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9226 * examples/test-auth.c:
9227 test: add permissions to auth test
9228 Ass some permissions to the media factory in the test.
9230 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9232 * gst/rtsp-server/rtsp-auth.c:
9233 * gst/rtsp-server/rtsp-auth.h:
9234 * gst/rtsp-server/rtsp-client.c:
9235 auth: simplify auth checks
9236 Remove client from methods, it's now in the state
9237 Perform the check specified by the string, use the information from the
9238 thread local context.
9240 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9242 * gst/rtsp-server/rtsp-client.c:
9243 * gst/rtsp-server/rtsp-client.h:
9244 client: add state to current thread
9245 Add the client to the ClientState object.
9246 Place the ClientState on the current thread.
9248 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9250 * gst/rtsp-server/rtsp-media-factory.c:
9251 * gst/rtsp-server/rtsp-media-factory.h:
9252 * gst/rtsp-server/rtsp-media.c:
9253 * gst/rtsp-server/rtsp-media.h:
9254 media: make it possible to set permissions
9255 Make it possible to set permissions on media and media factory objects
9257 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9259 * gst/rtsp-server/Makefile.am:
9260 * gst/rtsp-server/rtsp-permissions.c:
9261 * gst/rtsp-server/rtsp-permissions.h:
9262 permissions: add permissions object
9263 Add a mini object to store permissions based on a role.
9265 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9267 * examples/test-auth.c:
9268 * gst/rtsp-server/rtsp-auth.c:
9269 * gst/rtsp-server/rtsp-auth.h:
9270 * gst/rtsp-server/rtsp-client.c:
9271 auth: add auth checks
9272 Add an enum with auth checks and implement the checks in the auth object.
9273 Perform the checks from the client.
9275 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9277 * examples/test-auth.c:
9278 * gst/rtsp-server/rtsp-auth.c:
9279 * gst/rtsp-server/rtsp-auth.h:
9280 * gst/rtsp-server/rtsp-client.h:
9281 auth: use the token after authentication
9282 After we authenticated a user, keep the Token around in the state.
9284 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9286 * gst/rtsp-server/rtsp-client.c:
9287 * gst/rtsp-server/rtsp-media.c:
9288 * gst/rtsp-server/rtsp-media.h:
9289 * tests/check/gst/media.c:
9290 media: add optional context for bus messages
9291 Add an optional mainloop to _prepare that will handle the bus messages instead
9292 of always using the shared mainloop.
9294 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9296 * gst/rtsp-server/Makefile.am:
9297 * gst/rtsp-server/rtsp-token.c:
9298 * gst/rtsp-server/rtsp-token.h:
9299 token: add authorization token
9300 Add a simply miniobject that contains the authorizations. The object contains a
9301 GstStructure that hold all authorization fields. When a user is authenticated,
9302 the auth module will create a Token for the user. The token is then used to
9303 check what operations the user is allowed to do and various other configuration
9306 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * examples/test-auth.c:
9309 * gst/rtsp-server/rtsp-auth.c:
9310 * gst/rtsp-server/rtsp-auth.h:
9311 * gst/rtsp-server/rtsp-client.c:
9312 * gst/rtsp-server/rtsp-client.h:
9313 * gst/rtsp-server/rtsp-media-factory.c:
9314 * gst/rtsp-server/rtsp-media-factory.h:
9315 * gst/rtsp-server/rtsp-media.c:
9316 * gst/rtsp-server/rtsp-media.h:
9317 auth: remove auth from media and factory
9318 Remove the auth object from media and factory. We want to have the RTSPClient
9319 authenticate and authorize resources, there is no need to place another auth
9320 manager on the media/factory.
9322 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9324 * examples/test-auth.c:
9325 * gst/rtsp-server/rtsp-auth.c:
9326 * gst/rtsp-server/rtsp-auth.h:
9327 * gst/rtsp-server/rtsp-client.h:
9328 auth: add support for multiple basic auth tokens
9329 Make it possible to add multiple basic authorisation tokens to one authorization
9330 object. Associate with each token an authorization group that will define what
9331 capabilities are allowed.
9333 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * gst/rtsp-server/rtsp-client.c:
9336 client: error out on non-aggregate control
9337 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9339 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9341 * gst/rtsp-server/rtsp-client.c:
9342 client: rework setup request a little
9343 Cache the media in DESCRIBE based on the longest matching path with the uri
9344 that we can find in the mount points.
9345 Rework the setup request a little to get the media from the session or from
9346 the longest matching path, this way we can derive the control string as
9347 everything after the path instead of hardcoding it.
9348 Find the stream based on the control string and only open a session when all
9351 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9353 * gst/rtsp-server/rtsp-media.c:
9354 * gst/rtsp-server/rtsp-media.h:
9355 media: add method to find a stream by control url
9357 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9359 * gst/rtsp-server/rtsp-stream.c:
9360 * gst/rtsp-server/rtsp-stream.h:
9361 stream: add method to check control url of stream
9363 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9365 * gst/rtsp-server/rtsp-client.c:
9366 * gst/rtsp-server/rtsp-session-media.c:
9367 * gst/rtsp-server/rtsp-session-media.h:
9368 * gst/rtsp-server/rtsp-session.c:
9369 * gst/rtsp-server/rtsp-session.h:
9370 session: use path matching for session media
9371 Use a path string instead of a uri to lookup session media in the sessions. Also
9372 use path matching to find the largest possible path that matches.
9374 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9376 * gst/rtsp-server/rtsp-client.c:
9377 * gst/rtsp-server/rtsp-mount-points.c:
9378 * gst/rtsp-server/rtsp-mount-points.h:
9379 * tests/check/gst/mountpoints.c:
9380 mount-points: remove useless vmethod
9381 Making lookups in the mount points should not be done with a URL, if there is a
9382 mapping to be done from URL to mount points, we'll need to do it somewhere
9385 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9387 * gst/rtsp-server/rtsp-mount-points.c:
9388 * gst/rtsp-server/rtsp-mount-points.h:
9389 * tests/check/gst/mountpoints.c:
9390 mount-points: improve mount point searching
9391 Use a GSequence to keep track of the mount points.
9392 Match a URL to the longest matching registered mount point. This should be the
9393 URL to perform aggreagate control and the remainder is the stream specific
9395 Add some unit tests for this.
9397 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9399 * gst/rtsp-server/Makefile.am:
9400 rtsp-server: Allow building of static library
9402 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9404 * tests/check/gst/mediafactory.c:
9405 tests: fix compilation
9407 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9409 * gst/rtsp-server/rtsp-sdp.c:
9410 sdp: get control string from stream
9411 Use the control string as configured in the stream.
9413 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9415 * gst/rtsp-server/rtsp-stream.c:
9416 * gst/rtsp-server/rtsp-stream.h:
9417 stream: add methods and property to set control string
9419 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9421 * gst/rtsp-server/rtsp-client.c:
9423 Rename variables for clarity
9424 Keep media in state when we can
9426 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9428 * gst/rtsp-server/rtsp-client.c:
9429 * gst/rtsp-server/rtsp-stream.c:
9430 * gst/rtsp-server/rtsp-stream.h:
9431 stream: add more support for IPv6
9432 Rename _get_address to _get_multicast_address in GstRTSPStream to
9433 make it clear that this function only deals with multicast.
9434 Make it possible to have both an IPv4 and IPv6 multicast address on
9435 a stream. Give the client an IPv4 or IPv6 address depending on the
9436 address it used to connect to the server.
9437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9439 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9441 * gst/rtsp-server/rtsp-client.c:
9444 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9446 * gst/rtsp-server/rtsp-stream.c:
9447 stream: handle failed port allocation
9448 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9449 can't allocate any family at all. Also keep track of what port families we
9451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9453 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9455 * gst/rtsp-server/rtsp-stream.c:
9456 stream: improve docs
9458 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9460 * gst/rtsp-server/rtsp-stream-transport.c:
9461 stream-transport: remove old if 0 block
9463 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9465 * tests/check/gst/client.c:
9467 gst_rtsp_client_get_uri() has been removed
9468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9470 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9472 * gst/rtsp-server/rtsp-client.c:
9473 * gst/rtsp-server/rtsp-client.h:
9474 client: add method to filter managed sessions
9475 Add a method to filter the sessions managed by this client connection.
9476 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9478 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9480 * gst/rtsp-server/rtsp-client.c:
9481 * gst/rtsp-server/rtsp-client.h:
9482 client: remove _get_uri() method
9483 Remove the get_uri() method on the client. A client has no uri, the uri
9484 property is an internal property to manage the last cached media for
9487 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9489 * gst/rtsp-server/rtsp-media-factory.h:
9490 media-factory: fix typo
9492 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9494 * gst/rtsp-server/rtsp-media.c:
9495 rtsp-media: Do not leak the query in default_query_stop
9496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9498 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9500 * gst/rtsp-server/rtsp-media.c:
9501 media: don't unlock when conversion fails
9502 Don't unlock the state lock when conversion fails because it was not locked.
9504 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9506 * gst/rtsp-server/rtsp-media.c:
9507 * gst/rtsp-server/rtsp-media.h:
9508 Add query_position and query_stop vmethods to rtsp-media
9510 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9512 * gst/rtsp-server/rtsp-media.c:
9513 Fix typo in property install for rtsp-media's time-provider
9515 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9517 * gst/rtsp-server/rtsp-client.c:
9518 * gst/rtsp-server/rtsp-client.h:
9519 client: clean some variables
9520 Clean some variables and add some guards to _send_request()
9522 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9524 * gst/rtsp-server/rtsp-client.c:
9525 * gst/rtsp-server/rtsp-client.h:
9526 Add gst_rtsp_client_send_request API
9527 This makes it possible to send arbitrary messages to a client, such as
9528 SET_PARAMETER or GET_PARAMETER
9530 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9532 * gst/rtsp-server/rtsp-media.c:
9533 * gst/rtsp-server/rtsp-media.h:
9534 media: add _get_element() method
9535 Add method to get the element used when creating the media.
9536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9538 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9540 * gst/rtsp-server/rtsp-media.c:
9543 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9545 * gst/rtsp-server/rtsp-stream.c:
9546 * gst/rtsp-server/rtsp-stream.h:
9547 stream: allow access to the rtp session
9548 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9550 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9552 * gst/rtsp-server/rtsp-stream.c:
9553 * gst/rtsp-server/rtsp-stream.h:
9554 dscp qos support in gst-rtsp-stream
9555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9557 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9559 * tests/check/gst/rtspserver.c:
9561 Actually do what the comment says. Also keep the old code around, not sure what
9562 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9563 it currently doesn't.
9565 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9567 * gst/rtsp-server/rtsp-client.c:
9568 client: also watch newly created session
9569 When we newly created a session, start watching it immediately instead of
9570 on the next request.
9572 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9574 * tests/check/gst/client.c:
9575 tests: add unit test for new-session
9576 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9578 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9580 * gst/rtsp-server/rtsp-client.c:
9581 client: emit new-session when new session is created
9582 Only emit new-session when we created a new session for a client, not when a
9583 client picked up a previous session.
9584 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9586 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9588 * gst/rtsp-server/rtsp-client.c:
9589 client: handle asterisk as path in requests
9590 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9592 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9594 * gst/rtsp-server/rtsp-media.c:
9595 media: handle segment query format mismatch
9596 It's possible that the segment query returns with a different format than what
9597 we asked for, handle this case also.
9599 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9601 * gst/rtsp-server/rtsp-media.c:
9602 media: use segment stop in collect_media_stats
9603 Use segment stop instead of duration as range end point.
9604 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9606 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9608 * gst/rtsp-server/rtsp-media.c:
9609 * tests/check/gst/media.c:
9610 rtsp-media: Do not leak the element in take_pipeline
9611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9613 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9615 * gst/rtsp-server/rtsp-client.c:
9616 * gst/rtsp-server/rtsp-client.h:
9617 rtsp-client: Make configure_client_transport virtual
9618 This patch makes configure_client_transport virtual. The functionality is
9619 needed to handle some weird clients sending multicast transport settings as url
9621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9623 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9625 * gst/rtsp-server/rtsp-client.c:
9626 * gst/rtsp-server/rtsp-client.h:
9627 rtsp-client: Make param_set and param_get virtual
9628 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9630 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9632 * gst/rtsp-server/rtsp-client.c:
9633 * gst/rtsp-server/rtsp-media.c:
9634 * gst/rtsp-server/rtsp-media.h:
9635 media: convert_range replaces get_range_times
9636 get_range_times worked for handling UTC ranges for seeks, but we also
9637 need to convert back from NPT to the requested unit in
9638 get_range_string. convert_range is now used for both.
9639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9641 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 * gst/rtsp-server/rtsp-client.c:
9644 * gst/rtsp-server/rtsp-sdp.c:
9645 * gst/rtsp-server/rtsp-sdp.h:
9646 sdp: cleanup sdp info
9647 We don't need to pass the proto, we can more easily check a boolean.
9648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9650 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9652 * gst/rtsp-server/rtsp-sdp.c:
9653 use 0.0.0.0 or :: for c= line instead of server address
9655 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9657 * gst/rtsp-server/rtsp-client.c:
9658 use local address, not remote, in SDP
9659 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9661 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9664 Automatic update of common submodule
9665 From 098c0d7 to 01a7a46
9667 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9669 * gst/rtsp-server/rtsp-media.c:
9670 * gst/rtsp-server/rtsp-media.h:
9671 media: possibility to override range time conversion
9672 Make it possible to override the conversion from GstRTSPTimeRange to
9673 GstClockTimes, that is done before seeking on the media
9674 pipeline. Overriding can be useful for UTC ranges, where the default
9675 conversion gives nanoseconds since 1900.
9676 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9678 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9680 * gst/rtsp-server/rtsp-server.c:
9681 * gst/rtsp-server/rtsp-server.h:
9682 rtsp-server: Expose the use_client_settings API
9683 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9685 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9687 * gst/rtsp-server/rtsp-client.c:
9688 * gst/rtsp-server/rtsp-stream.c:
9689 * gst/rtsp-server/rtsp-stream.h:
9690 rtspstream: handle both ipv4 and ipv6 clients
9691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9693 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9695 * gst/rtsp-server/rtsp-sdp.c:
9696 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9697 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9698 We already have a way to place extra attributes in the SDP by using a string
9699 property with prefix x- or a- in the caps.
9701 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9703 * gst/rtsp-server/rtsp-sdp.c:
9704 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9705 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9706 We already have a way to place extra attributes in the SDP, just make a string
9707 property in the payloader with a- or x- prefix.
9709 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9711 * gst/rtsp-server/rtsp-sdp.c:
9712 rtsp: place a- and x- properties as attributes
9713 application/x-rtp has properties with a- and x- prefixes that should be
9714 placed as attributes in the SDP for the media instead of being added to the
9717 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9719 * examples/Makefile.am:
9720 * examples/test-video.c:
9721 example: add TLS example
9723 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9725 * gst/rtsp-server/rtsp-server.c:
9726 * gst/rtsp-server/rtsp-server.h:
9727 server: add support for TLS
9728 Add methods to set and get a TLS certificate.
9729 Add vmethod to configure a new connection. By default, configure the TLS
9730 certificate in a new connection if needed.
9732 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9734 * gst/rtsp-server/rtsp-server.c:
9735 * gst/rtsp-server/rtsp-server.h:
9736 server: remove accept_client vmethod
9737 This vmethod is not very useful so remove it.
9739 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9741 * gst/rtsp-server/rtsp-server.c:
9742 server: don't crash on NULL GError
9744 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
9746 * gst/rtsp-server/rtsp-session-pool.c:
9747 rtsp-session-pool: corrected session timeout detection
9748 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
9750 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9752 * gst/rtsp-server/rtsp-client.c:
9753 client: improve debug
9755 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9757 * gst/rtsp-server/rtsp-client.c:
9758 * gst/rtsp-server/rtsp-client.h:
9759 * gst/rtsp-server/rtsp-server.c:
9760 server: refactor connection setup
9761 Let the server accept the socket connection and construct a GstRTSPConnection
9762 from it. Remove the code from the client and let the client only deal with
9763 a fully configure GstRTSPConnection object.
9764 We will need this later when the server will configure the connection for
9767 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9769 * gst/rtsp-server/rtsp-stream.c:
9770 stream: keep the transport object alive
9771 Keep the transport object alive while we have it as qdata on the
9774 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
9776 * gst/rtsp-server/rtsp-client.c:
9777 * gst/rtsp-server/rtsp-server.c:
9778 rtsp-server: Do not crash on nmapping of server
9779 * generate error when gst_rtsp_connection_accept fails
9780 * do not stop accepting incoming connections because
9781 accepting a client fails
9782 https://bugzilla.gnome.org/show_bug.cgi?id=701072
9784 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
9786 * gst/rtsp-server/rtsp-client.c:
9787 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
9788 https://bugzilla.gnome.org/show_bug.cgi?id=700953
9790 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9792 * gst/rtsp-server/rtsp-sdp.c:
9793 rtsp-sdp: Parse framerate caps field and set SDP attribute
9794 The SDP attribute and its format is described in RFC4566.
9795 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9797 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
9799 * gst/rtsp-server/rtsp-sdp.c:
9800 rtsp-sdp: Parse width/height from caps and set SDP attribute
9801 The SDP attribute and its format is described in RFC6064.
9802 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9804 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
9806 * gst/rtsp-server/rtsp-sdp.c:
9807 * tests/check/gst/client.c:
9808 rtsp-sdp: add bandwidth line
9809 https://bugzilla.gnome.org/show_bug.cgi?id=699220
9811 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9814 Automatic update of common submodule
9815 From 5edcd85 to 098c0d7
9817 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9819 * tests/check/gst/media.c:
9820 tests: add dynamic payloader prepare/unprepare check
9822 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9824 * gst/rtsp-server/rtsp-media.c:
9825 media: release lock when removing fakesink
9827 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9829 * gst/rtsp-server/rtsp-stream.c:
9830 stream: set elements to NULL before removing
9831 When removing a stream, set the elements to NULL first. This avoids
9832 element-is-not-in-NULL-state errors when we dispose the elements.
9834 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9837 Automatic update of common submodule
9838 From 3cb3d3c to 5edcd85
9840 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9842 * gst/rtsp-server/rtsp-media.c:
9843 * gst/rtsp-server/rtsp-media.h:
9844 media: listen to pad-removed signals
9845 Listen to the pad-removed signal and remove the stream associated with the
9847 Add signal to be notified of the removed pad.
9848 Remove the fakesink in unprepare()
9849 Fix signatures of the signal methods
9851 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9853 * examples/test-sdp.c:
9854 tests: add example of reusable pipelines
9856 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
9858 * gst/rtsp-server/rtsp-stream.c:
9859 * gst/rtsp-server/rtsp-stream.h:
9860 stream: add method to get the srcpad
9862 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9864 * tests/check/gst/media.c:
9865 check: add media prepare/unprepare test
9866 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9868 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
9870 * gst/rtsp-server/rtsp-media.c:
9871 media: disconnect from signal handlers in unprepare()
9872 We connected to the pad-added and no-more-pads signals in prepare() so
9873 we need to disconnect from them in unprepare().
9874 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9876 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
9878 * gst/rtsp-server/rtsp-media.c:
9879 media: don't free streams array
9880 Don't free the streams array in the unprepare() method, they were not
9882 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9884 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
9886 * gst/rtsp-server/rtsp-media.c:
9887 media: don't unref the pipeline in unprepare
9888 Unprepare() should undo what prepare() does. Because the pipeline is
9889 not created in prepare(), we should not unref it in unprepare()
9891 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
9893 * gst/rtsp-server/rtsp-stream.c:
9894 stream: clear session and caps for reuse
9895 Set the session and caps to NULL after unref otherwise we might unref
9897 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9899 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
9901 * gst/rtsp-server/rtsp-client.c:
9902 client: send out teardown signal before tearing down
9903 The advantage is that in the signal handler you get direct access to
9904 information about what streams are about to get torn down (in the
9905 GstRTSPClientState).
9906 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
9908 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
9910 * gst/rtsp-server/rtsp-client.c:
9911 * gst/rtsp-server/rtsp-client.h:
9912 client: expose connection
9913 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
9915 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
9918 Automatic update of common submodule
9919 From aed87ae to 3cb3d3c
9921 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9923 * gst/rtsp-server/rtsp-media.c:
9924 * gst/rtsp-server/rtsp-media.h:
9925 * gst/rtsp-server/rtsp-session-media.c:
9926 * gst/rtsp-server/rtsp-session-media.h:
9927 media: add method to get the base_time of the pipeline
9928 Together with a shared clock, this base-time could eventually be sent to
9929 the client so that it can reconstruct the exact running-time of the clock
9932 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9934 * gst/rtsp-server/Makefile.am:
9935 * gst/rtsp-server/rtsp-media.c:
9936 * gst/rtsp-server/rtsp-media.h:
9937 * gst/rtsp-server/rtsp-sdp.c:
9938 media: add GstNetTimeProvider support
9939 Add a property to let the media provide a GstNetTimeProvider for its clock.
9940 Make methods to get the clock and nettimeprovider
9941 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
9942 provider and also the current time of the clock. This should make it possible
9943 for (GStreamer) clients to slave their clock to the server clock.
9945 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
9948 Automatic update of common submodule
9949 From 04c7a1e to aed87ae
9951 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9953 * gst/rtsp-server/rtsp-media.c:
9954 media: wait for buffering to complete
9955 Wait for buffering to complete before changing the state to the target state.
9957 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9959 * gst/rtsp-server/rtsp-media.c:
9960 media: small cleanup
9962 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
9964 * tests/check/gst/rtspserver.c:
9965 tests: remove extra unref in test_setup_non_existing_stream
9966 The unref is not needed anymore, teardown runs without it.
9967 https://bugzilla.gnome.org/show_bug.cgi?id=696542
9969 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
9971 * tests/check/gst/rtspserver.c:
9972 tests: GSocketService cleanup in test_bind_already_in_use
9973 Use g_socket_service_stop so the rtspserver test stops listening for
9974 incoming connections in test_bind_already_in_use.
9975 https://bugzilla.gnome.org/show_bug.cgi?id=696541
9977 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
9979 * gst/rtsp-server/rtsp-media-factory.c:
9980 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
9981 Instead use a GWeakRef which is safe to use
9982 This is a known GLib bug, see:
9983 https://bugzilla.gnome.org/show_bug.cgi?id=667145
9985 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
9987 * gst/rtsp-server/rtsp-client.c:
9988 * gst/rtsp-server/rtsp-media.c:
9989 * gst/rtsp-server/rtsp-media.h:
9990 * gst/rtsp-server/rtsp-sdp.c:
9991 * tests/check/gst/media.c:
9992 * tests/check/gst/rtspserver.c:
9993 rtsp-media/client: Reply to PLAY request with same type of Range
9994 Remember the type of Range from the PLAY request and use the same type for
9997 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
9999 * gst/rtsp-server/rtsp-client.c:
10000 * gst/rtsp-server/rtsp-client.h:
10001 * tests/check/gst/client.c:
10002 rtsp-client: expose uri
10004 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
10006 * tests/check/gst/mediafactory.c:
10007 tests: Hold ref while creating second media
10008 To test if the media aren't shared, make sure we keep the first one while creating a second
10009 otherwise the same memory address may be reused.
10011 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
10014 configure: remove out-of-date comment
10016 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10019 .gitignore: ignore more build files
10021 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10023 * tests/check/Makefile.am:
10024 tests: use right _LIBS variable for gst-plugins-base libs
10026 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10028 * tests/check/Makefile.am:
10029 check: add librtp to libs
10031 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10033 * tests/check/gst/rtspserver.c:
10034 tests: Add test to check selecting a port the server will send from
10036 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10038 * tests/check/gst/rtspserver.c:
10039 tests: Make sure packets are actually received
10041 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10043 * gst/rtsp-server/rtsp-stream.c:
10044 stream: Select unicast address from pool if appropriate
10046 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10048 * gst/rtsp-server/rtsp-stream.c:
10049 stream: Properties are always there in Gst 1.0
10051 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10053 * tests/check/gst/addresspool.c:
10054 tests: Add tests for unicast addresses in pool
10056 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10058 * gst/rtsp-server/rtsp-address-pool.c:
10059 * tests/check/gst/addresspool.c:
10060 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10062 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10064 * docs/libs/gst-rtsp-server-sections.txt:
10065 * gst/rtsp-server/rtsp-address-pool.c:
10066 * gst/rtsp-server/rtsp-address-pool.h:
10067 * gst/rtsp-server/rtsp-stream.c:
10068 * tests/check/gst/addresspool.c:
10069 address-pool: Add unicast addresses
10071 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10074 * gst/rtsp-server/rtsp-server.c:
10075 * tests/check/gst/rtspserver.c:
10076 rtsp-server: Limit the number of threads per server instance
10077 If we exceed the maximum, just round robin the clients over the existing
10080 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10082 * gst/rtsp-server/rtsp-server.c:
10083 rtsp-server: No need to store the GMainContext in the client context
10085 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10087 * tests/check/gst/rtspserver.c:
10088 tests: Add test for client disconnection
10090 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10092 * tests/check/gst/rtspserver.c:
10093 tests: Test client and session timeouts with multiple threads
10095 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10097 * gst/rtsp-server/rtsp-address-pool.c:
10098 * gst/rtsp-server/rtsp-auth.c:
10099 * gst/rtsp-server/rtsp-client.c:
10100 * gst/rtsp-server/rtsp-media-factory-uri.c:
10101 * gst/rtsp-server/rtsp-media-factory.c:
10102 * gst/rtsp-server/rtsp-media.c:
10103 * gst/rtsp-server/rtsp-mount-points.c:
10104 * gst/rtsp-server/rtsp-server.c:
10105 * gst/rtsp-server/rtsp-session-media.c:
10106 * gst/rtsp-server/rtsp-session-pool.c:
10107 * gst/rtsp-server/rtsp-session.c:
10108 Document locking and its order
10110 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10112 * tests/check/gst/rtspserver.c:
10113 tests: Test that slow DESCRIBE don't block other clients
10115 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10117 * tests/check/gst/client.c:
10118 tests: Add tests for client-requested multicast address
10120 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10122 * docs/libs/gst-rtsp-server-sections.txt:
10123 docs: Put the various functions in the right sections
10125 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10127 * docs/libs/gst-rtsp-server-docs.sgml:
10128 * docs/libs/gst-rtsp-server-sections.txt:
10129 * gst/rtsp-server/rtsp-address-pool.c:
10130 * gst/rtsp-server/rtsp-address-pool.h:
10131 docs: Generate docs for GstRTSPAddressPool
10133 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10135 * gst/rtsp-server/rtsp-client.c:
10136 * gst/rtsp-server/rtsp-stream.c:
10137 * gst/rtsp-server/rtsp-stream.h:
10138 client: Check client provided addresses against the address pool
10140 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10142 * gst/rtsp-server/rtsp-address-pool.c:
10143 * gst/rtsp-server/rtsp-address-pool.h:
10144 * tests/check/gst/addresspool.c:
10145 address-pool: Add API to request a specific address from the pool
10146 Also add relevant unit tests.
10148 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10150 * tests/check/gst/mediafactory.c:
10151 tests: Check the passing around of a RTSPAddressPool
10152 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10153 way down to the stream.
10155 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10157 * tests/check/gst/addresspool.c:
10158 tests: Add more tests for the address pool
10160 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10162 * gst/rtsp-server/rtsp-address-pool.c:
10163 address-pool: Fix off by one error
10164 When splitting a port range, the port after a skip is not part of range.
10166 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10169 Automatic update of common submodule
10170 From 2de221c to 04c7a1e
10172 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10175 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10176 AM_CONFIG_HEADER was removed in automake 1.13
10177 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10179 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10182 Automatic update of common submodule
10183 From a942293 to 2de221c
10185 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10187 * gst/rtsp-server/rtsp-client.c:
10188 client: make sure the watch exists while sending data
10189 Protect the send_func with a lock. This allows us to wait for sending
10190 to complete before changing the send_func and user_data. We add an
10191 extra ref to the watch to make sure that it remains valid during
10193 When closing the connection, set the send_func to NULL
10194 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10196 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10198 * tests/check/Makefile.am:
10199 tests: use GST_*_1_0 environment variables everywhere
10200 The _1_0 suffixed environment variables override the
10201 non-suffixed ones, so if we're in an environment that
10202 sets the _1_0 suffixed ones, such as jhbuild, we need
10203 to set those to make sure ours actually always get
10206 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10209 Automatic update of common submodule
10210 From acb04d9 to a942293
10212 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10214 * gst/rtsp-server/rtsp-client.c:
10215 rtsp-client: set the client backlog
10216 Set the client backlog to a reasonable default
10218 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10220 * gst/rtsp-server/rtsp-media.c:
10221 rtsp-media: Make the element a constructor parameter
10222 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10224 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10226 * docs/libs/Makefile.am:
10227 docs: Link with gcov library when gcov is enabled
10228 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10230 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10232 * gst/rtsp-server/rtsp-media.c:
10233 media: match prepare with unprepare
10234 Really unprepare when there were an equal amount of prepare calls.
10236 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10238 * gst/rtsp-server/rtsp-media.c:
10239 media: media has to be unprepared in finalize
10240 Because unprepare takes away the last ref on the media.
10242 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10244 * gst/rtsp-server/rtsp-client.c:
10245 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10246 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10247 We can't use the refcount to trigger unprepare because it is the unprepare call
10248 that removes the last refcount after all messages are consumed. What we should
10249 probably do is make a prepared refcount and only unprepare when the refcount
10252 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10254 * gst/rtsp-server/rtsp-media.c:
10255 media: let the source unref the last media ref
10256 the last ref to the media is held by the source so we don't need to add more ref
10257 and unrefs, we simply destroy the media when the source is gone.
10259 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10261 * gst/rtsp-server/rtsp-media.c:
10262 media: improve debug
10264 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10266 * gst/rtsp-server/rtsp-media.c:
10268 Make sure we are in the right state when collecting the position and duration.
10269 Only make ourselves PREPARED when we were previously PREPARING.
10271 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10273 * gst/rtsp-server/rtsp-media.c:
10274 media: use g_object_ref/unref for GObjects
10276 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10278 * gst/rtsp-server/rtsp-client.c:
10279 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10280 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10281 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10282 isn't being used anymore.
10284 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10286 * gst/rtsp-server/rtsp-media.c:
10287 Fix compiler warning
10289 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10291 * gst/rtsp-server/rtsp-media-factory-uri.c:
10292 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10294 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10296 * gst/rtsp-server/rtsp-session-media.h:
10299 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10301 * gst/rtsp-server/rtsp-media.c:
10302 * tests/check/gst/media.c:
10303 media: avoid element leak
10305 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10307 * gst/rtsp-server/rtsp-media.c:
10308 media: require an element in media constructor
10310 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10312 * gst/rtsp-server/rtsp-client.c:
10313 Revert "client: TEARDOWN brings that state to Init again"
10314 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10315 The object is already disposed, there is no point in setting the state.
10317 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10319 * gst/rtsp-server/rtsp-client.c:
10320 client: TEARDOWN brings that state to Init again
10322 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10324 * docs/libs/gst-rtsp-server-sections.txt:
10325 * examples/test-auth.c:
10326 * gst/rtsp-server/rtsp-auth.c:
10327 * gst/rtsp-server/rtsp-auth.h:
10328 * gst/rtsp-server/rtsp-client.c:
10329 * gst/rtsp-server/rtsp-client.h:
10330 * gst/rtsp-server/rtsp-media-factory-uri.c:
10331 * gst/rtsp-server/rtsp-media-factory-uri.h:
10332 * gst/rtsp-server/rtsp-media-factory.c:
10333 * gst/rtsp-server/rtsp-media-factory.h:
10334 * gst/rtsp-server/rtsp-media.c:
10335 * gst/rtsp-server/rtsp-media.h:
10336 * gst/rtsp-server/rtsp-mount-points.c:
10337 * gst/rtsp-server/rtsp-mount-points.h:
10338 * gst/rtsp-server/rtsp-sdp.c:
10339 * gst/rtsp-server/rtsp-server.c:
10340 * gst/rtsp-server/rtsp-server.h:
10341 * gst/rtsp-server/rtsp-session-media.c:
10342 * gst/rtsp-server/rtsp-session-media.h:
10343 * gst/rtsp-server/rtsp-session-pool.c:
10344 * gst/rtsp-server/rtsp-session-pool.h:
10345 * gst/rtsp-server/rtsp-session.c:
10346 * gst/rtsp-server/rtsp-session.h:
10347 * gst/rtsp-server/rtsp-stream-transport.c:
10348 * gst/rtsp-server/rtsp-stream-transport.h:
10349 * gst/rtsp-server/rtsp-stream.c:
10350 * gst/rtsp-server/rtsp-stream.h:
10351 * tests/check/gst/media.c:
10352 rtsp: make object details private
10353 Make all object details private
10354 Add methods to access private bits
10356 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10358 * tests/check/Makefile.am:
10359 * tests/check/gst/media.c:
10360 tests: add media tests
10362 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10364 * gst/rtsp-server/rtsp-media.c:
10365 media: check if prepared for some methods
10366 Check that the media object is prepared before doing seek and getting the
10367 current position etc.
10368 Add some g_return checks.
10370 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10372 * tests/check/Makefile.am:
10373 * tests/check/gst/mediafactory.c:
10374 tests: add mediafactory test
10376 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10378 * gst/rtsp-server/rtsp-stream.c:
10379 stream: improve debug
10381 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10383 * gst/rtsp-server/rtsp-media.c:
10384 * gst/rtsp-server/rtsp-media.h:
10385 media: unref pipeline in finalize to avoid leaking it
10387 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10389 * gst/rtsp-server/rtsp-media-factory-uri.c:
10390 * gst/rtsp-server/rtsp-media.c:
10391 rtsp: use gst_object_unref on GstObjects
10393 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10395 * gst/rtsp-server/rtsp-media-factory.c:
10396 media-factory: require an url
10398 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10400 * examples/test-uri.c:
10401 examples: fix include
10403 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10405 * gst/rtsp-server/rtsp-server.h:
10406 server: remove unused include
10408 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10410 * tests/check/Makefile.am:
10411 * tests/check/gst/mountpoints.c:
10412 tests: add test for mountpoints
10414 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10416 * gst/rtsp-server/rtsp-client.c:
10417 client: fix factory leak
10418 Keep the factory in the state object only for authorization checks and make
10419 sure we unref it on failure. Also don't keep invalid objects in the state
10422 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10424 * gst/rtsp-server/rtsp-mount-points.c:
10425 mounts: add g_return_if guards
10427 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10429 * tests/check/gst/client.c:
10430 tests: add more tests
10432 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10434 * gst/rtsp-server/rtsp-client.c:
10435 client: improve debug
10437 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10439 * gst/rtsp-server/rtsp-client.c:
10440 client: improve debug and fix leaks
10441 Cleanup the uri and session when there is a bad request.
10443 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10448 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10450 * tests/check/gst/client.c:
10451 test: add test for session in options request
10453 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10455 * gst/rtsp-server/rtsp-client.c:
10456 client: use 454 when session can't be found
10457 We should use 454 when a session can't be found because there was no session
10458 pool configured in the server. This is not a server configuration problem
10459 because the server on which the request is done might not be the same one that
10460 will keep the sessions for us and so it does not need to support sessions.
10462 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10464 * gst/rtsp-server/rtsp-client.c:
10465 client: only free connection when there is one
10466 It's possible that the client doesn't have a connection when we try to free it.
10468 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10470 * tests/check/Makefile.am:
10471 * tests/check/gst/client.c:
10472 tests: add unit test for the client object
10474 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10476 * gst/rtsp-server/rtsp-client.c:
10477 client: small cleanup
10479 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10481 * gst/rtsp-server/rtsp-client.h:
10482 client: remove unused include
10484 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10486 * gst/rtsp-server/rtsp-client.c:
10487 client: fix compilation
10489 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10491 * gst/rtsp-server/rtsp-client.c:
10492 client: call destroy without the lock
10494 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10496 * gst/rtsp-server/rtsp-client.c:
10497 * gst/rtsp-server/rtsp-client.h:
10498 client: make the client usable without a socket
10499 Make a method to let the client handle a message and a callback when the client
10500 wants us to send a response message back. This makes it possible to also use the
10501 client object without the sockets, which should make it easier to test.
10503 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10505 * gst/rtsp-server/rtsp-client.c:
10506 * gst/rtsp-server/rtsp-client.h:
10507 client: small cleanup
10509 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10511 * docs/libs/gst-rtsp-server-sections.txt:
10512 * gst/rtsp-server/rtsp-client.c:
10513 * gst/rtsp-server/rtsp-client.h:
10514 * gst/rtsp-server/rtsp-server.c:
10515 client: remove reference to server
10516 We don't need to keep a ref to the server
10518 2012-11-26 16:30:16 +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 locking
10523 Also add some g_return_if()
10525 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10527 * gst/rtsp-server/rtsp-client.c:
10528 client: log more errors
10530 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10532 * gst/rtsp-server/rtsp-client.c:
10533 client: fix compilation
10535 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10537 * gst/rtsp-server/rtsp-client.c:
10538 * gst/rtsp-server/rtsp-client.h:
10539 client: add generic close-after-send support
10540 Add a property to send_response() to close the connection after the response has
10541 been sent to the client.
10543 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10546 * docs/libs/gst-rtsp-server-docs.sgml:
10547 * docs/libs/gst-rtsp-server-sections.txt:
10548 * docs/libs/gst-rtsp-server.types:
10549 * examples/test-auth.c:
10550 * examples/test-launch.c:
10551 * examples/test-mp4.c:
10552 * examples/test-multicast.c:
10553 * examples/test-multicast2.c:
10554 * examples/test-ogg.c:
10555 * examples/test-readme.c:
10556 * examples/test-sdp.c:
10557 * examples/test-uri.c:
10558 * examples/test-video.c:
10559 * gst/rtsp-server/Makefile.am:
10560 * gst/rtsp-server/rtsp-auth.h:
10561 * gst/rtsp-server/rtsp-client.c:
10562 * gst/rtsp-server/rtsp-client.h:
10563 * gst/rtsp-server/rtsp-media-mapping.c:
10564 * gst/rtsp-server/rtsp-media-mapping.h:
10565 * gst/rtsp-server/rtsp-mount-points.c:
10566 * gst/rtsp-server/rtsp-mount-points.h:
10567 * gst/rtsp-server/rtsp-server.c:
10568 * gst/rtsp-server/rtsp-server.h:
10569 * gst/rtsp-server/rtsp-session-media.c:
10570 * gst/rtsp-server/rtsp-session-pool.c:
10571 * gst/rtsp-server/rtsp-session-pool.h:
10572 * tests/check/gst/rtspserver.c:
10573 MediaMapping -> MountPoints
10574 Describes better what the object manages.
10576 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10579 configure: bump required version of -base
10581 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10583 * gst/rtsp-server/rtsp-media.c:
10586 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10588 * gst/rtsp-server/rtsp-media.c:
10589 * gst/rtsp-server/rtsp-media.h:
10590 media: support more Range formats
10591 Use the new -base methods to convert the Range string into a seek start and stop
10594 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10596 * examples/test-launch.c:
10597 examples: fix whitespace
10599 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10601 * examples/test-auth.c:
10602 test-auth: add example of how to remove sessions
10603 Add an example of the session filter api.
10605 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * examples/test-uri.c:
10608 test-uri: remove mapping example
10610 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10612 * examples/test-uri.c:
10613 test-uri: fix callback signature
10615 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10617 * gst/rtsp-server/rtsp-media-factory.c:
10618 factory: keep ref to factory while media active
10619 While the media from a factory is alive, keep a ref to the factory.
10620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10622 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10624 * gst/rtsp-server/rtsp-media-factory-uri.c:
10625 factory-uri: add some debug
10627 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10629 * gst/rtsp-server/rtsp-stream.c:
10630 stream: set udp sources to PLAYING
10631 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10632 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10634 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10636 * gst/rtsp-server/rtsp-media-factory-uri.c:
10637 factory-uri: take ref to factory
10638 Take a ref to the factory that we place in our list.
10640 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10642 * tests/Makefile.am:
10643 * tests/test-reuse.c:
10644 test: add test for server reuse
10645 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10647 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10649 * gst/rtsp-server/rtsp-server.c:
10650 server: start and stop multiple times
10651 Stop listening on the RTSP port when the GSource is removed, so clients
10652 can't connect and the server can be started again.
10653 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10655 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10657 * gst/rtsp-server/rtsp-server.c:
10658 server: fix small leak
10660 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10662 * gst/rtsp-server/rtsp-media.c:
10663 media: unref source in finish_unprepare
10664 The source is created in prepare, unref it in finish_unprepare.
10665 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10667 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10669 * gst/rtsp-server/rtsp-client.c:
10670 * gst/rtsp-server/rtsp-media.c:
10671 rtsp-media: remove bus watch before finalizing
10672 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10673 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10674 the GDestroyNotify function.
10675 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10676 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10677 gst_rtsp_media_unprepare before unreffing the media.
10678 This way, the bus watch will be removed before the media is finalized.
10679 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10681 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10683 * gst/rtsp-server/rtsp-client.c:
10684 * gst/rtsp-server/rtsp-client.h:
10685 client: wait until the TEARDOWN response is sent to close the connection
10686 Responses can be sent async so we need to wait until the TEARDOWN response has
10687 been written before we close the connection to the client. This avoids the risk
10688 of writing/polling closed sockets.
10689 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10691 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10693 * gst/rtsp-server/rtsp-stream.c:
10694 rtsp-stream: plug socket leak
10695 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10697 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10700 Automatic update of common submodule
10701 From 6bb6951 to a72faea
10703 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10705 * gst/rtsp-server/rtsp-media-factory-uri.c:
10706 rtsp-server: don't use deprecated API
10708 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10710 * gst/rtsp-server/rtsp-client.c:
10711 rtsp-client: fix unused-but-set-variable compiler warning
10712 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10714 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10717 * docs/libs/gst-rtsp-server-sections.txt:
10718 * gst/rtsp-server/rtsp-client.c:
10721 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10723 * examples/Makefile.am:
10724 * examples/test-multicast2.c:
10725 examples: add another multicast example
10726 Add an example for how to configure separate multicast ranges for each media
10729 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10731 * examples/test-multicast.c:
10734 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10736 * gst/rtsp-server/rtsp-client.c:
10737 * gst/rtsp-server/rtsp-media.c:
10738 * gst/rtsp-server/rtsp-session-media.c:
10739 * gst/rtsp-server/rtsp-session-media.h:
10740 * gst/rtsp-server/rtsp-stream-transport.c:
10741 * gst/rtsp-server/rtsp-stream-transport.h:
10742 stream: use the address managed by the stream
10743 Use the address managed by the stream for multicast. This allows us to have 1
10744 multicast address for each stream.
10745 Because the address is now managed by the stream we don't have to pass it around
10747 Set the address pool on the streams.
10749 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10751 * gst/rtsp-server/rtsp-client.c:
10752 * gst/rtsp-server/rtsp-media.c:
10753 * gst/rtsp-server/rtsp-stream.c:
10754 rtsp: improve debug
10756 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10758 * gst/rtsp-server/rtsp-media.c:
10759 * gst/rtsp-server/rtsp-media.h:
10760 media: add signal for new streams
10761 This allows applications to listen for new streams and configure properties on
10762 them, like the address pool.
10764 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10766 * gst/rtsp-server/rtsp-media.c:
10767 media: configure address pool in new streams
10769 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10771 * gst/rtsp-server/rtsp-stream.c:
10772 * gst/rtsp-server/rtsp-stream.h:
10773 stream: add methods to deal with address pool
10774 Add methods to get and set the address pool for the stream
10775 Add method to allocate and get the multicast addresses for this stream.
10777 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10779 * docs/libs/gst-rtsp-server-sections.txt:
10780 * gst/rtsp-server/rtsp-media.c:
10781 * gst/rtsp-server/rtsp-media.h:
10782 media: remove MTU property
10783 It is a stream property
10785 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10787 * gst/rtsp-server/rtsp-client.c:
10788 client: set blocksize only on stream
10789 Set the blocksize only on the current stream.
10791 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10793 * gst/rtsp-server/rtsp-stream.c:
10794 stream: share src and sink sockets
10795 the allocated socket is in the used-socket property, not socket.
10797 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10799 * gst/rtsp-server/rtsp-address-pool.c:
10800 * gst/rtsp-server/rtsp-address-pool.h:
10801 * gst/rtsp-server/rtsp-client.c:
10802 * gst/rtsp-server/rtsp-session-media.c:
10803 * gst/rtsp-server/rtsp-session-media.h:
10804 * gst/rtsp-server/rtsp-stream-transport.c:
10805 * gst/rtsp-server/rtsp-stream-transport.h:
10806 * tests/check/gst/addresspool.c:
10807 rtsp: make address-pool return an address object
10808 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
10809 store more info in the structure and allows us to more easily return the address
10810 to the right pool when no longer needed.
10811 Pass the address to the StreamTransport so that we can return it to the pool
10812 when the stream transport is freed or changed.
10814 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10816 * examples/Makefile.am:
10817 * examples/test-multicast.c:
10818 examples: add multicast example
10819 Show how to set up the multicast address pool so that media can be
10820 server with multicast.
10822 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10824 * gst/rtsp-server/rtsp-client.c:
10825 * gst/rtsp-server/rtsp-media-factory.c:
10826 * gst/rtsp-server/rtsp-media-factory.h:
10827 * gst/rtsp-server/rtsp-media.c:
10828 * gst/rtsp-server/rtsp-media.h:
10829 rtsp: use AddressPool
10830 Remove the multicast_group property.
10831 Use the configured addresspool to allocate multicast addresses.
10833 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10835 * gst/rtsp-server/rtsp-address-pool.c:
10836 * gst/rtsp-server/rtsp-address-pool.h:
10837 address-pool: add clear method
10839 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10841 * gst/rtsp-server/rtsp-address-pool.c:
10842 address-pool: small cleanups
10844 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10846 * tests/check/Makefile.am:
10847 * tests/check/gst/addresspool.c:
10848 tests: add addresspool unit test
10850 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10852 * gst/rtsp-server/Makefile.am:
10853 * gst/rtsp-server/rtsp-address-pool.c:
10854 * gst/rtsp-server/rtsp-address-pool.h:
10855 address-pool: add object to manage multicast addresses
10856 Make an object that can manage a rage of multicast addresses and ports.
10858 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10860 * gst/rtsp-server/rtsp-server.c:
10861 server: set default max-threads property
10863 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10865 * gst/rtsp-server/rtsp-media.c:
10866 media: wait for concurrent _prepare
10867 If a prepare is busy, wait for the result.
10869 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10871 * gst/rtsp-server/rtsp-media.c:
10872 media: add lock around message handler
10873 We don't want to dispatch messages while we are still processing the result of
10876 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10878 * gst/rtsp-server/rtsp-media.c:
10879 * gst/rtsp-server/rtsp-media.h:
10880 media: add lock to protect state changes
10882 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10884 * gst/rtsp-server/rtsp-stream.c:
10885 * gst/rtsp-server/rtsp-stream.h:
10886 stream: add locking
10888 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10890 * gst/rtsp-server/rtsp-stream-transport.c:
10891 * gst/rtsp-server/rtsp-stream-transport.h:
10892 * gst/rtsp-server/rtsp-stream.c:
10893 stream-transport: add keep-alive method
10895 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10897 * gst/rtsp-server/rtsp-stream-transport.c:
10898 * gst/rtsp-server/rtsp-stream-transport.h:
10899 * gst/rtsp-server/rtsp-stream.c:
10900 stream-transport: add method to handle RTP/RTCP
10901 Call new methods instead of poking into the structures directly.
10903 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10905 * gst/rtsp-server/rtsp-session-media.c:
10906 * gst/rtsp-server/rtsp-session-media.h:
10907 session-media: add locking
10909 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10911 * gst/rtsp-server/rtsp-session.c:
10912 * gst/rtsp-server/rtsp-session.h:
10913 session: add locking
10915 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10917 * gst/rtsp-server/rtsp-server.c:
10918 server: free old socket
10920 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10922 * gst/rtsp-server/rtsp-media-mapping.c:
10923 * gst/rtsp-server/rtsp-media-mapping.h:
10924 mapping: add locking
10926 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10928 * gst/rtsp-server/rtsp-media-factory.c:
10929 media-factory: add locking
10931 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10933 * gst/rtsp-server/rtsp-auth.c:
10934 * gst/rtsp-server/rtsp-auth.h:
10937 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10939 * gst/rtsp-server/rtsp-server.c:
10940 * gst/rtsp-server/rtsp-server.h:
10941 server: add max-thread property
10943 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10945 * gst/rtsp-server/rtsp-server.c:
10946 * gst/rtsp-server/rtsp-server.h:
10947 server: use a threadpool for the mainloops
10949 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10951 * gst/rtsp-server/rtsp-client.c:
10952 * gst/rtsp-server/rtsp-client.h:
10953 client: rename method
10954 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
10955 don't really create the client from the socket, we use the socket for the
10958 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10960 * gst/rtsp-server/rtsp-client.c:
10961 * gst/rtsp-server/rtsp-client.h:
10962 * gst/rtsp-server/rtsp-server.c:
10963 server: rework maincontext handling in clients
10964 Make a separate method to attach a client to a MainContext.
10965 Let the server decide in what GMainContext the client will operate and give this
10966 context to the client in attach. Then the server can later decide to use a
10967 separate thread for each client or just use the mainthread.
10969 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10971 * gst/rtsp-server/rtsp-client.c:
10972 * gst/rtsp-server/rtsp-session.c:
10973 * gst/rtsp-server/rtsp-session.h:
10974 session: move session header code in session object
10976 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10980 * examples/test-auth.c:
10981 * examples/test-launch.c:
10982 * examples/test-mp4.c:
10983 * examples/test-ogg.c:
10984 * examples/test-readme.c:
10985 * examples/test-sdp.c:
10986 * examples/test-uri.c:
10987 * examples/test-video.c:
10988 * gst/rtsp-server/rtsp-auth.c:
10989 * gst/rtsp-server/rtsp-auth.h:
10990 * gst/rtsp-server/rtsp-client.c:
10991 * gst/rtsp-server/rtsp-client.h:
10992 * gst/rtsp-server/rtsp-media-factory-uri.c:
10993 * gst/rtsp-server/rtsp-media-factory-uri.h:
10994 * gst/rtsp-server/rtsp-media-factory.c:
10995 * gst/rtsp-server/rtsp-media-factory.h:
10996 * gst/rtsp-server/rtsp-media-mapping.c:
10997 * gst/rtsp-server/rtsp-media-mapping.h:
10998 * gst/rtsp-server/rtsp-media.c:
10999 * gst/rtsp-server/rtsp-media.h:
11000 * gst/rtsp-server/rtsp-params.c:
11001 * gst/rtsp-server/rtsp-params.h:
11002 * gst/rtsp-server/rtsp-sdp.c:
11003 * gst/rtsp-server/rtsp-sdp.h:
11004 * gst/rtsp-server/rtsp-server.c:
11005 * gst/rtsp-server/rtsp-server.h:
11006 * gst/rtsp-server/rtsp-session-media.c:
11007 * gst/rtsp-server/rtsp-session-media.h:
11008 * gst/rtsp-server/rtsp-session-pool.c:
11009 * gst/rtsp-server/rtsp-session-pool.h:
11010 * gst/rtsp-server/rtsp-session.c:
11011 * gst/rtsp-server/rtsp-session.h:
11012 * gst/rtsp-server/rtsp-stream-transport.c:
11013 * gst/rtsp-server/rtsp-stream-transport.h:
11014 * gst/rtsp-server/rtsp-stream.c:
11015 * gst/rtsp-server/rtsp-stream.h:
11016 * tests/check/gst/rtspserver.c:
11017 * tests/test-cleanup.c:
11020 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11022 * gst/rtsp-server/rtsp-media.c:
11023 * gst/rtsp-server/rtsp-session-media.c:
11024 * gst/rtsp-server/rtsp-session.c:
11025 rtsp-server: added annotations to indicate type of ownership transfer of return values
11026 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11028 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11031 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11033 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11036 * bindings/Makefile.am:
11037 * bindings/vala/Makefile.am:
11038 * bindings/vala/gst-rtsp-server-0.10.deps:
11039 * bindings/vala/gst-rtsp-server-0.10.vapi:
11040 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11041 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11042 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11043 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11044 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11046 bindings: remove vala bindings
11047 They'll be reunited with the other GStreamer bindings
11048 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11050 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11052 * gst/rtsp-server/rtsp-client.c:
11053 * gst/rtsp-server/rtsp-session-media.c:
11054 * gst/rtsp-server/rtsp-session-media.h:
11055 * gst/rtsp-server/rtsp-stream-transport.c:
11056 * gst/rtsp-server/rtsp-stream-transport.h:
11057 rtsp: only create transport when needed
11058 Only create the StreamTransport when configured.
11060 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11062 * gst/rtsp-server/rtsp-client.c:
11063 client: small cleanup
11065 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11067 * gst/rtsp-server/rtsp-client.c:
11068 * gst/rtsp-server/rtsp-client.h:
11069 * gst/rtsp-server/rtsp-stream-transport.c:
11070 * gst/rtsp-server/rtsp-stream-transport.h:
11071 rtsp: refactor configuration of transport
11072 Move the configuration of the transport to a place where it makes
11075 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11077 * gst/rtsp-server/rtsp-client.c:
11078 client: refactor transport parsing
11080 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11082 * gst/rtsp-server/rtsp-client.c:
11083 client: refuse to change the MTU on shared media
11084 If we change the MTU of chared media, it changes for all clients.
11085 We don't want to set the MTU to something large for clients that
11088 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11090 * examples/test-mp4.c:
11091 * gst/rtsp-server/rtsp-media.c:
11092 small fixes to docs and debug
11094 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11096 * gst/rtsp-server/rtsp-stream.c:
11097 stream: transports must already have been removed
11099 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11101 * gst/rtsp-server/rtsp-media.c:
11102 * gst/rtsp-server/rtsp-stream.c:
11103 * gst/rtsp-server/rtsp-stream.h:
11104 stream: improve join and leave of the pipeline
11106 Do the cleanup properly
11109 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11111 * gst/rtsp-server/rtsp-media.c:
11112 media: move unprepare below default implementation
11113 Makes it easier to find the default implementation
11115 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11117 * gst/rtsp-server/rtsp-media.c:
11118 media: signal unprepared when we actually finish
11120 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11122 * gst/rtsp-server/rtsp-media.c:
11123 media: no need to unlock, unprepare does that when needed
11125 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11127 * docs/libs/gst-rtsp-server-sections.txt:
11128 * gst/rtsp-server/rtsp-media-factory.h:
11129 * gst/rtsp-server/rtsp-media-mapping.c:
11130 * gst/rtsp-server/rtsp-media.h:
11131 * gst/rtsp-server/rtsp-params.c:
11132 * gst/rtsp-server/rtsp-server.c:
11133 * gst/rtsp-server/rtsp-session-pool.h:
11134 * gst/rtsp-server/rtsp-session.c:
11135 * gst/rtsp-server/rtsp-session.h:
11136 * gst/rtsp-server/rtsp-stream-transport.h:
11137 * gst/rtsp-server/rtsp-stream.h:
11140 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11142 * gst/rtsp-server/rtsp-client.c:
11143 * gst/rtsp-server/rtsp-media-mapping.h:
11144 * gst/rtsp-server/rtsp-media.c:
11145 * gst/rtsp-server/rtsp-media.h:
11146 * gst/rtsp-server/rtsp-server.h:
11147 * gst/rtsp-server/rtsp-stream.c:
11148 * gst/rtsp-server/rtsp-stream.h:
11149 rtsp: fix MTU setting
11150 Fix setting of the MTU. There is no need for a vmethod.
11152 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11157 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11160 configure: bump version number after refactoring
11162 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11164 * gst/rtsp-server/Makefile.am:
11165 * gst/rtsp-server/rtsp-client.c:
11166 * gst/rtsp-server/rtsp-client.h:
11167 * gst/rtsp-server/rtsp-media-factory-uri.c:
11168 * gst/rtsp-server/rtsp-media-factory.c:
11169 * gst/rtsp-server/rtsp-media-factory.h:
11170 * gst/rtsp-server/rtsp-media.c:
11171 * gst/rtsp-server/rtsp-media.h:
11172 * gst/rtsp-server/rtsp-sdp.c:
11173 * gst/rtsp-server/rtsp-session-media.c:
11174 * gst/rtsp-server/rtsp-session-media.h:
11175 * gst/rtsp-server/rtsp-session.c:
11176 * gst/rtsp-server/rtsp-session.h:
11177 * gst/rtsp-server/rtsp-stream-transport.c:
11178 * gst/rtsp-server/rtsp-stream-transport.h:
11179 * gst/rtsp-server/rtsp-stream.c:
11180 * gst/rtsp-server/rtsp-stream.h:
11181 rtsp: massive refactoring
11182 Make GObjects from the remaining simple structures.
11183 Remove GstRTSPSessionStream, it's not needed.
11184 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11185 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11186 a GstRTSPStream should be transported to a client.
11187 Rename GstRTSPMediaFactory::get_element -> create_element because that
11188 more accurately describes what it does.
11189 Make nice methods instead of poking in the structures.
11190 Move some methods inside the relevant object source code.
11191 Use GPtrArray to store objects instead of plain arrays, it is more
11192 natural and allows us to more easily clean up.
11193 Move the allocation of udp ports to the Stream object. The Stream object
11194 contains the elements needed to stream the media to a client.
11195 Improve the prepare and unprepare methods. Unprepare should now undo
11196 everything prepare did. Improve also async unprepare when doing EOS on
11197 shutdown. Make sure we always unprepare correctly.
11199 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11201 * gst/rtsp-server/rtsp-client.c:
11202 rtsp-client: Unref server address clients connected to
11203 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11205 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11207 * gst/rtsp-server/rtsp-server.c:
11208 rtsp-server: don't ref server socket if it is NULL
11209 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11210 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11212 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11214 * tests/check/Makefile.am:
11215 tests: Add libgio link dependency
11216 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11218 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11220 * gst/rtsp-server/rtsp-media-mapping.c:
11221 * gst/rtsp-server/rtsp-media-mapping.h:
11222 rtsp-media-mapping: rename find_media vfunc to find_factory
11223 The virtual method and class method should have the same name
11224 so it is correctly represented in GIR file
11225 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11227 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11229 * gst/rtsp-server/rtsp-auth.c:
11230 * gst/rtsp-server/rtsp-client.c:
11231 * gst/rtsp-server/rtsp-media-factory-uri.c:
11232 * gst/rtsp-server/rtsp-media-factory.c:
11233 * gst/rtsp-server/rtsp-media-mapping.c:
11234 * gst/rtsp-server/rtsp-media.c:
11235 * gst/rtsp-server/rtsp-server.c:
11236 * gst/rtsp-server/rtsp-session-pool.c:
11237 * gst/rtsp-server/rtsp-session.c:
11238 rtsp-server: fixed comments and GIR annotations
11239 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11241 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11243 * gst/rtsp-server/rtsp-media-mapping.c:
11244 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11246 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11248 * gst/rtsp-server/rtsp-server.c:
11249 rtsp-server: allow binding on port 0 (binds on a random port)
11251 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11253 * gst/rtsp-server/rtsp-server.c:
11254 * gst/rtsp-server/rtsp-server.h:
11255 rtsp-server: add bound-port property
11256 bound-port can be used to retrieve the port number when the server is bound on
11257 port 0, which binds on a random port.
11259 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11261 * gst/rtsp-server/rtsp-media-factory.c:
11262 * gst/rtsp-server/rtsp-media-factory.h:
11263 rtsp-media-factory: make ::get_element overridable by GI bindings
11264 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11265 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11266 as the invoker for ::get_element(), making it overridable by GI generated
11269 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11271 * gst/rtsp-server/rtsp-media-factory-uri.c:
11272 rtsp-media-factory-uri: don't autoplug parsers in a loop
11273 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11276 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11278 * gst/rtsp-server/Makefile.am:
11279 Explicitly link against gio. Fix link error on mac.
11281 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11283 * gst/rtsp-server/rtsp-session.c:
11284 session: add ttl to the transport header in SETUP
11285 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11287 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11289 * gst/rtsp-server/rtsp-client.c:
11290 * gst/rtsp-server/rtsp-client.h:
11291 * gst/rtsp-server/rtsp-media.c:
11292 client: Use client transport settings for multicast if allowed.
11293 This patch makes it possible for the client to send transport settings for
11294 multicast (destination && ttl). Client settings must be explicitly allowed or
11295 the server will use its own settings.
11296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11298 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11301 Automatic update of common submodule
11302 From 6c0b52c to 6bb6951
11304 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11306 * gst/rtsp-server/rtsp-client.c:
11307 rtsp-client: do not destroy the rtsp watch
11308 Don't destroy the client watch while dispatching. The rtsp watch is
11309 automatically destroyed after the rtsp watch function closed() has
11311 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11313 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11316 Automatic update of common submodule
11317 From 4f962f7 to 6c0b52c
11319 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11321 * gst/rtsp-server/rtsp-media.c:
11322 media: fix check for seekability
11324 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11326 * gst/rtsp-server/rtsp-client.c:
11327 client: use more GIO
11328 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11330 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11332 * gst/rtsp-server/rtsp-server.c:
11333 server: remove obsolete includes
11335 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11337 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11338 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11339 be available in "on_new_ssrc". The transports are added in
11340 gst_rtsp_media_set_state when going to PLAYING state. However,
11341 "on_new_ssrc" might be called before this happens.
11342 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11344 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11346 * gst/rtsp-server/rtsp-client.c:
11347 * gst/rtsp-server/rtsp-client.h:
11348 rtsp-client: add signals for rtsp requests (fixes #683287)
11350 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11352 * gst/rtsp-server/rtsp-client.c:
11353 * gst/rtsp-server/rtsp-client.h:
11354 add new-session signal to rtsp-client (fixes #683058)
11356 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11359 Automatic update of common submodule
11360 From 668acee to 4f962f7
11362 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11364 * gst/rtsp-server/rtsp-server.c:
11365 * tests/check/gst/rtspserver.c:
11366 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11367 Do not assume that *error is set in g_socket_address_enumerator_next.
11368 Added test_bind_already_in_use unit-test.
11369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11371 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11374 Automatic update of common submodule
11375 From 94ccf4c to 668acee
11377 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11379 * gst/rtsp-server/rtsp-client.c:
11380 * gst/rtsp-server/rtsp-client.h:
11381 rtsp-client: make create_sdp virtual method
11382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11384 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11387 Automatic update of common submodule
11388 From 98e386f to 94ccf4c
11390 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11392 * gst/rtsp-server/rtsp-client.c:
11395 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11397 * gst/rtsp-server/rtsp-client.c:
11398 * gst/rtsp-server/rtsp-client.h:
11399 * gst/rtsp-server/rtsp-server.c:
11400 * gst/rtsp-server/rtsp-server.h:
11401 rtsp-server: use an existing socket to establish HTTP tunnel
11402 Make it possible to transfer a socket from an HTTP server to be used as
11403 an RTSP over HTTP tunnel.
11405 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11407 * gst/rtsp-server/rtsp-client.c:
11408 * gst/rtsp-server/rtsp-media.c:
11409 * gst/rtsp-server/rtsp-media.h:
11410 rtsp: Handle the blocksize parameter
11411 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11413 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11415 * tests/check/Makefile.am:
11416 * tests/check/gst/rtspserver.c:
11417 Have unit test get header from source dir, not installed dir
11418 This makes compilation of unit tests work in a build directory other
11419 than the source directory.
11420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11422 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11424 * gst/rtsp-server/rtsp-media.c:
11425 rtsp-media: update for gst_element_make_from_uri() changes
11427 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11430 * tests/Makefile.am:
11431 * tests/check/Makefile.am:
11432 * tests/check/gst/rtspserver.c:
11433 rtsp: add unit test
11434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11436 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11438 * gst/rtsp-server/rtsp-media.c:
11439 rtsp-media: don't collect media stats when going to NULL
11440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11442 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11444 * gst/rtsp-server/rtsp-client.c:
11445 client: don't leak transports
11447 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11449 * gst/rtsp-server/rtsp-client.c:
11450 rtsp-client: free transport on no_stream in SETUP handler
11452 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11454 * gst/rtsp-server/rtsp-client.c:
11455 rtsp-client: changed session media iteration
11456 In client_unlink_session: now don't iterate in session->medias
11457 list where items are removed by gst_rtsp_session_release_media.
11458 Instead, repeatedly remove the first item.
11460 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11462 * gst/rtsp-server/rtsp-client.c:
11463 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11464 GstRTSPSessionMedia is not a GObject type. When the
11465 GstRTSPSession is freed, it will free the media.
11467 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11469 * gst/rtsp-server/rtsp-media-factory.c:
11470 factory: plug pad leak in collect_streams
11471 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11472 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11473 will take one reference, and the other reference will otherwise
11474 give a memory leak.
11476 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11479 configure: suppress some warnings when debug is disabled
11480 Warnings about unused variables should be suppressed if core has the
11481 debug system disabled.
11482 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11484 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11486 * docs/libs/Makefile.am:
11487 docs: fix build in uninstalled setup
11488 Include gst-plugins-base libs properly.
11490 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11492 * docs/libs/gst-rtsp-server.types:
11493 docs: include headers defining rtsp-server object types
11494 Fixes compiler warnings during docs build.
11495 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11497 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11500 configure: Add warning flags for compiler when configuring
11501 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11503 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11506 Automatic update of common submodule
11507 From 03a0e57 to 98e386f
11509 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11512 Automatic update of common submodule
11513 From 1fab359 to 03a0e57
11515 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11517 * gst/rtsp-server/rtsp-client.c:
11518 client: fix GSocketAddress leak in gst_rtsp_client_accept
11519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11521 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11524 Automatic update of common submodule
11525 From f1b5a96 to 1fab359
11527 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11530 Automatic update of common submodule
11531 From 92b7266 to f1b5a96
11533 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11536 Automatic update of common submodule
11537 From ec1c4a8 to 92b7266
11539 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11542 Automatic update of common submodule
11543 From 3429ba6 to ec1c4a8
11545 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11547 * gst/rtsp-server/rtsp-auth.c:
11548 * gst/rtsp-server/rtsp-client.c:
11549 * gst/rtsp-server/rtsp-media-factory-uri.c:
11550 * gst/rtsp-server/rtsp-server.c:
11551 rtsp: fix compiler warnings
11552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11554 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11557 Automatic update of common submodule
11558 From dc70203 to 3429ba6
11560 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11562 * gst/rtsp-server/rtsp-client.c:
11563 * gst/rtsp-server/rtsp-media-factory.c:
11564 * gst/rtsp-server/rtsp-media-factory.h:
11565 * gst/rtsp-server/rtsp-media.c:
11566 * gst/rtsp-server/rtsp-media.h:
11567 * gst/rtsp-server/rtsp-server.c:
11568 * gst/rtsp-server/rtsp-server.h:
11569 * gst/rtsp-server/rtsp-session-pool.c:
11570 * gst/rtsp-server/rtsp-session-pool.h:
11571 rtsp-server: port to new thread API
11573 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11576 Automatic update of common submodule
11577 From 6db25be to dc70203
11579 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11581 * gst/rtsp-server/rtsp-auth.c:
11582 * gst/rtsp-server/rtsp-auth.h:
11583 * gst/rtsp-server/rtsp-client.c:
11584 rtsp-server: Fix compilation and compiler warnings
11586 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11590 * gst/rtsp-server/Makefile.am:
11591 configure: Modernize autotools setup a bit
11592 Also we now only create tar.bz2 and tar.xz tarballs.
11594 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11597 Automatic update of common submodule
11598 From 464fe15 to 6db25be
11600 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11603 Automatic update of common submodule
11604 From 7fda524 to 464fe15
11606 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11609 * docs/libs/Makefile.am:
11610 * docs/version.entities.in:
11611 * gst-rtsp.spec.in:
11612 * gst/rtsp-server/Makefile.am:
11613 * pkgconfig/Makefile.am:
11614 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11615 * pkgconfig/gstreamer-rtsp-server.pc.in:
11616 * tests/Makefile.am:
11617 rtsp-server: Update versioning
11619 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11621 Merge remote-tracking branch 'origin/0.10'
11623 gst/rtsp-server/rtsp-session-pool.c
11625 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11627 * gst/rtsp-server/rtsp-session-pool.c:
11628 rtsp-server: Don't use deprecated GLib API
11630 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11632 Replace master with 0.11
11634 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11636 Merge branch 'master' into 0.11
11638 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11640 Merge branch 'master' into 0.11
11642 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11645 A couple minor typo fixes
11647 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11649 * gst/rtsp-server/rtsp-media.c:
11650 media: fix state of the appqueue
11652 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11654 * gst/rtsp-server/rtsp-media-factory-uri.c:
11655 factory: use videoconvert
11657 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11659 * gst/rtsp-server/rtsp-media-factory-uri.c:
11660 factory: change to new style caps
11662 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11664 * gst/rtsp-server/rtsp-client.c:
11665 * gst/rtsp-server/rtsp-client.h:
11666 * gst/rtsp-server/rtsp-media-factory-uri.c:
11667 * gst/rtsp-server/rtsp-media.c:
11668 * gst/rtsp-server/rtsp-server.c:
11669 * gst/rtsp-server/rtsp-server.h:
11670 * gst/rtsp-server/rtsp-session-pool.c:
11671 rtsp-server: port to GIO
11674 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11677 configure: fix build
11679 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11682 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11683 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11685 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11688 * examples/Makefile.am:
11689 First rule of gst-rtsp-server club: don't talk about gst-phonon
11691 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11694 * pkgconfig/Makefile.am:
11695 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11696 * pkgconfig/gstreamer-rtsp-server.pc.in:
11697 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11698 For consistency with all other modules.
11700 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11702 * gst/rtsp-server/rtsp-client.c:
11703 rtsp-client: update for new map API
11705 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11708 * bindings/Makefile.am:
11709 * bindings/python/Makefile.am:
11710 * bindings/python/arg-types.py:
11711 * bindings/python/codegen/Makefile.am:
11712 * bindings/python/codegen/__init__.py:
11713 * bindings/python/codegen/argtypes.py:
11714 * bindings/python/codegen/code-coverage.py:
11715 * bindings/python/codegen/codegen.py:
11716 * bindings/python/codegen/definitions.py:
11717 * bindings/python/codegen/defsparser.py:
11718 * bindings/python/codegen/docextract.py:
11719 * bindings/python/codegen/docgen.py:
11720 * bindings/python/codegen/fileprefix.override:
11721 * bindings/python/codegen/fileprefixmodule.c:
11722 * bindings/python/codegen/h2def.py:
11723 * bindings/python/codegen/mergedefs.py:
11724 * bindings/python/codegen/mkskel.py:
11725 * bindings/python/codegen/override.py:
11726 * bindings/python/codegen/reversewrapper.py:
11727 * bindings/python/codegen/scmexpr.py:
11728 * bindings/python/rtspserver-types.defs:
11729 * bindings/python/rtspserver.defs:
11730 * bindings/python/rtspserver.override:
11731 * bindings/python/rtspservermodule.c:
11732 * bindings/python/test.py:
11734 python: remove pygst-based python bindings
11735 pygi is the future, apparently.
11737 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
11740 Automatic update of common submodule
11741 From c463bc0 to 7fda524
11743 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11746 Automatic update of common submodule
11747 From 2a59016 to c463bc0
11749 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11752 Automatic update of common submodule
11753 From 0807187 to 2a59016
11755 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11758 Automatic update of common submodule
11759 From 11f0cd5 to 0807187
11761 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11763 * examples/test-auth.c:
11764 example: update for new caps
11766 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11768 * examples/test-video.c:
11769 * gst/rtsp-server/rtsp-client.c:
11770 * gst/rtsp-server/rtsp-media-factory-uri.c:
11771 * gst/rtsp-server/rtsp-media.c:
11772 * gst/rtsp-server/rtsp-media.h:
11773 * gst/rtsp-server/rtsp-session.c:
11774 * gst/rtsp-server/rtsp-session.h:
11775 rtsp-server: port some more to 0.11
11777 Remove bufferlist stuff
11778 Update for new API.
11779 Add queue before appsink now that preroll-queue-len is gone.
11780 Update for request pad changes.
11782 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11784 Merge branch 'master' into 0.11
11786 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11788 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11789 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11790 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11792 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11794 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11795 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11796 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11798 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11800 Merge branch 'master' into 0.11
11802 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11804 * gst/rtsp-server/rtsp-media.c:
11805 * gst/rtsp-server/rtsp-media.h:
11806 media: add a seekable boolean
11807 Maintain the seekable state with a new variable instead of reusing the
11810 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
11812 * gst/rtsp-server/rtsp-media.c:
11813 Disallow seek in live media
11815 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11817 Merge branch 'master' into 0.11
11819 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
11821 * gst/rtsp-server/rtsp-server.c:
11822 #ifdef statements for windows socket creation were missing
11824 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
11827 Automatic update of common submodule
11828 From a39eb83 to 11f0cd5
11830 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
11833 Automatic update of common submodule
11834 From 605cd9a to a39eb83
11836 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11838 Merge branch 'master' into 0.11
11840 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11842 * gst/rtsp-server/rtsp-client.c:
11843 client: use method to access property
11845 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11847 * gst/rtsp-server/rtsp-media-factory.c:
11848 * gst/rtsp-server/rtsp-media-factory.h:
11849 media-factory: add protocols property
11850 Add a property to configure the allowed protocols in the media created from the
11853 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11855 * gst/rtsp-server/rtsp-media-factory.c:
11856 * gst/rtsp-server/rtsp-media-factory.h:
11857 media-factory: add media-configure signal
11858 Add signal to allow the application to configure the media after it was created
11861 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11863 * gst/rtsp-server/rtsp-client.c:
11864 client: use method to access property
11866 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11868 * gst/rtsp-server/rtsp-media-factory.c:
11869 * gst/rtsp-server/rtsp-media-factory.h:
11870 media-factory: add protocols property
11871 Add a property to configure the allowed protocols in the media created from the
11874 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11876 * gst/rtsp-server/rtsp-media-factory.c:
11877 * gst/rtsp-server/rtsp-media-factory.h:
11878 media-factory: add media-configure signal
11879 Add signal to allow the application to configure the media after it was created
11882 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11884 Merge branch 'master' into 0.11
11886 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11888 * gst/rtsp-server/rtsp-client.c:
11889 client: use media multicast group
11891 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11893 * gst/rtsp-server/rtsp-media-factory.h:
11894 * gst/rtsp-server/rtsp-server.h:
11895 * gst/rtsp-server/rtsp-session-pool.h:
11896 * gst/rtsp-server/rtsp-session.h:
11899 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11901 * gst/rtsp-server/rtsp-client.c:
11902 * gst/rtsp-server/rtsp-sdp.h:
11903 sdp: copy and free the server ip address
11904 Copy and free the server ip address to make memory management easier later.
11906 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11908 * gst/rtsp-server/rtsp-media-factory.c:
11909 media-factory: configure multicast in media
11911 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11913 * gst/rtsp-server/rtsp-media.c:
11914 * gst/rtsp-server/rtsp-media.h:
11915 media: add property for multicast group
11916 Add a property to configure the multicast group in the media.
11917 Based on patches from Marc Leeman and Robert Krakora.
11919 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11921 * gst/rtsp-server/rtsp-media-factory.c:
11922 * gst/rtsp-server/rtsp-media-factory.h:
11923 media-factory: add property for multicast group
11924 Add a property to configure the multicast group in the media factory.
11925 Based on patches from Marc Leeman and Robert Krakora.
11927 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11929 * gst/rtsp-server/rtsp-client.c:
11930 client: do configuration of transport in one place
11931 Move the configuration of the transport destination address to where we also
11932 configure the other bits.
11934 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11936 * gst/rtsp-server/rtsp-client.c:
11937 client: use media multicast group
11939 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11941 * gst/rtsp-server/rtsp-media-factory.h:
11942 * gst/rtsp-server/rtsp-server.h:
11943 * gst/rtsp-server/rtsp-session-pool.h:
11944 * gst/rtsp-server/rtsp-session.h:
11947 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11949 * gst/rtsp-server/rtsp-client.c:
11950 * gst/rtsp-server/rtsp-sdp.h:
11951 sdp: copy and free the server ip address
11952 Copy and free the server ip address to make memory management easier later.
11954 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11956 * gst/rtsp-server/rtsp-media-factory.c:
11957 media-factory: configure multicast in media
11959 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11961 * gst/rtsp-server/rtsp-media.c:
11962 * gst/rtsp-server/rtsp-media.h:
11963 media: add property for multicast group
11964 Add a property to configure the multicast group in the media.
11965 Based on patches from Marc Leeman and Robert Krakora.
11967 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11969 * gst/rtsp-server/rtsp-media-factory.c:
11970 * gst/rtsp-server/rtsp-media-factory.h:
11971 media-factory: add property for multicast group
11972 Add a property to configure the multicast group in the media factory.
11973 Based on patches from Marc Leeman and Robert Krakora.
11975 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11977 * gst/rtsp-server/rtsp-client.c:
11978 client: do configuration of transport in one place
11979 Move the configuration of the transport destination address to where we also
11980 configure the other bits.
11982 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11984 Merge branch 'master' into 0.11
11986 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11988 * gst/rtsp-server/rtsp-client.c:
11989 client: destroy pipeline on client disconnect with no prior TEARDOWN.
11990 The problem occurs when the client abruptly closes the connection without
11991 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
11992 server is where the pipeline gets torn down. Since this handler is not called,
11993 the pipeline remains and is up and running. Subsequent clients get their own
11994 pipelines and if the do not issue TEARDOWNs then those pipelines will also
11995 remain up and running. This is a resource leak.
11997 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11999 Merge branch 'master' into 0.11
12001 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
12003 * gst/rtsp-server/rtsp-media-factory.c:
12004 * gst/rtsp-server/rtsp-media-factory.h:
12005 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
12006 For example, it can be used to retrieve source elements like appsrc, in a more
12007 convenient way than subclassing get_element.
12009 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12011 Merge branch 'master' into 0.11
12013 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
12015 * gst/rtsp-server/rtsp-server.c:
12016 rtsp-server: hold on to reference while using object
12018 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12020 * gst/rtsp-server/rtsp-media.c:
12023 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12026 configure: use unstable api
12028 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12030 * gst/rtsp-server/rtsp-client.c:
12031 client: fix reference counting
12033 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12035 * gst/rtsp-server/rtsp-client.c:
12036 * gst/rtsp-server/rtsp-media.c:
12037 fix compiler warnings about unused variables
12039 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12041 * examples/test-launch.c:
12042 * examples/test-readme.c:
12043 * examples/test-uri.c:
12044 * examples/test-video.c:
12045 examples: tell rtsp uri when ready
12047 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12050 Automatic update of common submodule
12051 From 69b981f to 605cd9a
12053 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12055 * gst/rtsp-server/rtsp-client.c:
12056 client: update for buffer API change
12058 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12060 * gst/rtsp-server/Makefile.am:
12061 Makefile.am: 0.10 => @GST_MAJORMINOR@
12063 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12065 * gst/rtsp-server/rtsp-media-factory-uri.c:
12066 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12068 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12070 * gst/rtsp-server/.gitignore:
12071 .gitignore: 0.10 => 0.11
12073 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12075 * gst/rtsp-server/Makefile.am:
12076 Makefile.am: 0.10 => @GST_MAJORMINOR@
12078 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12080 Merge branch 'master' into 0.11
12082 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12085 Automatic update of common submodule
12086 From 9e5bbd5 to 69b981f
12088 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12091 Automatic update of common submodule
12092 From fd35073 to 9e5bbd5
12094 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12097 Automatic update of common submodule
12098 From 46dfcea to fd35073
12100 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12102 * gst/rtsp-server/rtsp-media-factory-uri.c:
12103 * gst/rtsp-server/rtsp-media.c:
12104 media: port to new caps API
12106 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12108 Merge branch 'master' into 0.11
12110 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12112 * bindings/vala/gst-rtsp-server-0.10.vapi:
12113 Updated Vala bindings.
12114 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12116 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12118 * gst/rtsp-server/rtsp-server.c:
12119 * gst/rtsp-server/rtsp-server.h:
12120 Add a signal for newly connected clients.
12121 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12123 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12125 * bindings/python/rtspserver.override:
12126 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12128 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12130 * gst/rtsp-server/Makefile.am:
12131 * gst/rtsp-server/rtsp-client.c:
12132 * gst/rtsp-server/rtsp-funnel.c:
12133 * gst/rtsp-server/rtsp-funnel.h:
12134 * gst/rtsp-server/rtsp-media.c:
12135 rtsp-server: port to 0.11
12137 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12142 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12144 Merge branch 'master' into 0.11
12149 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12152 Automatic update of common submodule
12153 From c3cafe1 to 46dfcea
12155 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12157 * bindings/python/Makefile.am:
12158 * bindings/python/rtspserver.defs:
12159 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12161 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12163 * bindings/python/arg-types.py:
12164 python bindings: add GstRTSPUrlParam
12165 Needed to implement MediaFactory virtual proxies
12167 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12169 * bindings/python/arg-types.py:
12170 python bindings: fix returning GstRTSPUrl types
12172 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12174 * bindings/python/arg-types.py:
12175 python bindings: add arg type for GstRTSPUrl
12177 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12179 * bindings/python/rtspserver.defs:
12180 python bindings: fix the definition of MediaFactory.collect_stream
12182 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12185 Automatic update of common submodule
12186 From 1ccbe09 to c3cafe1
12188 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12191 Automatic update of common submodule
12192 From 193b717 to 1ccbe09
12194 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12197 Automatic update of common submodule
12198 From b77e2bf to 193b717
12200 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12203 build: Include lcov.mak to allow test coverage report generation
12205 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12208 Automatic update of common submodule
12209 From d8814b6 to b77e2bf
12211 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12214 Automatic update of common submodule
12215 From 6aaa286 to d8814b6
12217 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12220 Automatic update of common submodule
12221 From 6aec6b9 to 6aaa286
12223 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12226 autogen: wingo signed comment
12228 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12230 * gst/rtsp-server/rtsp-session-pool.c:
12231 session: use full charset for RTSP session ID
12232 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12233 session ID more difficult.
12234 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12236 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12238 * gst/rtsp-server/Makefile.am:
12239 rtsp-server: Don't install the funnel header
12241 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12244 Automatic update of common submodule
12245 From 1de7f6a to 6aec6b9
12247 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12250 configure: require core/base 0.10.31
12251 Needed at least for gst_plugin_feature_rank_compare_func().
12253 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12256 Automatic update of common submodule
12257 From f94d739 to 1de7f6a
12259 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12261 * gst/rtsp-server/rtsp-media.c:
12262 media: remove more unused code
12264 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12266 * gst/rtsp-server/rtsp-media.c:
12267 * gst/rtsp-server/rtsp-media.h:
12268 media: remove duplicate filtering
12269 Remove the duplicate filtering code now that we have a released -good version.
12270 Give a warning instead.
12272 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12274 * gst/rtsp-server/rtsp-media-factory.c:
12275 * gst/rtsp-server/rtsp-media.c:
12276 media: fix default buffer size
12278 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12280 * gst/rtsp-server/rtsp-media-factory.c:
12281 * gst/rtsp-server/rtsp-media-factory.h:
12282 media-factory: add property to configure the buffer-size
12283 Add a property to configure the kernel UDP buffer size.
12285 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12287 * gst/rtsp-server/rtsp-media.c:
12288 * gst/rtsp-server/rtsp-media.h:
12289 media: add property to configure kernel buffer sizes
12290 Add a property to configure the kernel UDP buffer size.
12292 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12295 configure: set PYGOBJECT_REQ before using it
12296 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12298 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12300 * docs/Makefile.am:
12301 docs: recursive into sub-directories on 'make upload'
12303 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12305 * docs/libs/gst-rtsp-server-docs.sgml:
12306 * docs/version.entities.in:
12307 docs: mention full version these docs are for, not just major-minor
12309 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12312 back to development
12314 === release 0.10.8 ===
12316 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12321 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12323 * gst/rtsp-server/rtsp-server.c:
12324 rtsp-server: clarify docs a little
12326 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12328 * gst/rtsp-server/rtsp-media.c:
12329 media: init debug category before starting thread
12331 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12333 * gst/rtsp-server/rtsp-auth.c:
12334 auth: add realm to make it more spec compliant
12336 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12338 * gst/rtsp-server/rtsp-server.c:
12339 * gst/rtsp-server/rtsp-server.h:
12340 server: add locking
12342 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12344 * examples/test-video.c:
12345 example: improve example docs a little
12347 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12349 * gst/rtsp-server/rtsp-server.c:
12350 server: ensure the watch has a ref to the server
12352 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12354 * gst/rtsp-server/rtsp-server.c:
12355 server: simpify channel function
12357 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12359 * gst/rtsp-server/rtsp-server.c:
12360 * gst/rtsp-server/rtsp-server.h:
12361 server: simplify management of channel and source
12362 We don't need to keep around the channel and source objects. Let the mainloop
12363 and the source manage the source and channel respectively.
12365 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12371 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12373 * tests/.gitignore:
12374 * tests/Makefile.am:
12375 * tests/test-cleanup.c:
12376 tests: add tests directory and cleanup test
12378 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12380 * gst/rtsp-server/rtsp-media-factory-uri.c:
12381 * gst/rtsp-server/rtsp-media-factory.c:
12382 * gst/rtsp-server/rtsp-media-mapping.c:
12383 * gst/rtsp-server/rtsp-media.c:
12384 * gst/rtsp-server/rtsp-session-pool.c:
12385 * gst/rtsp-server/rtsp-session.c:
12386 server: improve debugging in various objects
12388 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12390 * gst/rtsp-server/rtsp-server.c:
12391 server: chain up to the parent finalize
12393 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12395 * bindings/python/rtspserver-types.defs:
12396 * bindings/python/rtspserver.defs:
12397 * bindings/python/rtspserver.override:
12398 * bindings/python/test.py:
12399 gst-rtsp-server: update python bindings
12401 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12403 * gst/rtsp-server/rtsp-client.c:
12404 client: use the response from the clientstate
12405 Create the response object only once and store in the client state.
12406 Make all methods use the state response,
12408 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12410 * gst/rtsp-server/rtsp-server.c:
12411 server: use signal to keep track of clients
12412 Keep track of all the clients that the server creates and remove them when they
12413 fire the 'closed' signal.
12415 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12417 * gst/rtsp-server/rtsp-client.c:
12418 * gst/rtsp-server/rtsp-client.h:
12419 client: emit signal when closing
12421 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12423 * examples/.gitignore:
12424 * examples/Makefile.am:
12425 * examples/test-auth.c:
12426 * examples/test-video.c:
12427 * gst/rtsp-server/rtsp-auth.c:
12428 * gst/rtsp-server/rtsp-auth.h:
12429 * gst/rtsp-server/rtsp-client.c:
12430 * gst/rtsp-server/rtsp-media-factory.c:
12431 * gst/rtsp-server/rtsp-media.c:
12432 * gst/rtsp-server/rtsp-media.h:
12433 * gst/rtsp-server/rtsp-session-pool.h:
12434 * gst/rtsp-server/rtsp-session.h:
12435 media: enable per factory authorisations
12436 Allow for adding a GstRTSPAuth on the factory and media level and check
12437 permissions when accessing the factory.
12438 Add hints to the auth methods for future more fine grained authorisation.
12439 Add example application for per factory authentication.
12441 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12443 * gst/rtsp-server/rtsp-auth.c:
12444 * gst/rtsp-server/rtsp-auth.h:
12445 * gst/rtsp-server/rtsp-client.c:
12446 * gst/rtsp-server/rtsp-client.h:
12447 * gst/rtsp-server/rtsp-params.c:
12448 * gst/rtsp-server/rtsp-params.h:
12449 rtsp-server: Pass ClientState structure arround
12450 Pass the collected information for the ongoing request in a GstRTSPClientState
12451 structure that we can then pass around to simplify the method arguments. This
12452 will also be handy when we implement logging functionality.
12454 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12456 * gst/rtsp-server/rtsp-media-factory.c:
12457 * gst/rtsp-server/rtsp-media-factory.h:
12458 media-factory: add methods to configure authorisation
12460 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12462 * gst/rtsp-server/rtsp-client.c:
12463 client: unref auth in finalize
12465 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12467 * gst/rtsp-server/rtsp-server.c:
12468 server: unref auth in finalize
12470 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12472 * docs/libs/gst-rtsp-server-docs.sgml:
12473 * docs/libs/gst-rtsp-server-sections.txt:
12474 * docs/libs/gst-rtsp-server.types:
12475 docs: add more docs
12477 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12479 * gst/rtsp-server/rtsp-server.c:
12480 * gst/rtsp-server/rtsp-server.h:
12481 server: separate create and accept
12482 Create separate create and accept methods so that subclasses can create custom
12484 Configure the server in the client object and prepare for keeping track of
12487 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12489 * gst/rtsp-server/rtsp-client.c:
12490 * gst/rtsp-server/rtsp-client.h:
12491 client: add support for setting the server.
12492 Add support for keeping a ref to the server that started this client
12495 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12497 * gst/rtsp-server/rtsp-auth.c:
12498 auth: fix memleak and add some docs
12499 Fix a memleak of the basic auth token.
12500 Add docs for the helper function
12502 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12504 * gst/rtsp-server/rtsp-auth.c:
12505 * gst/rtsp-server/rtsp-auth.h:
12506 * gst/rtsp-server/rtsp-client.c:
12507 client: delegate setup of auth to the manager
12508 Delegate the configuration of the authentication tokens to the manager object
12511 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12513 * examples/test-video.c:
12514 * gst/rtsp-server/Makefile.am:
12515 * gst/rtsp-server/rtsp-auth.c:
12516 * gst/rtsp-server/rtsp-auth.h:
12517 * gst/rtsp-server/rtsp-client.c:
12518 * gst/rtsp-server/rtsp-client.h:
12519 * gst/rtsp-server/rtsp-server.c:
12520 * gst/rtsp-server/rtsp-server.h:
12521 auth: add authentication object
12522 Add an object that can check the authorization of requests.
12523 Implement basic authentication.
12524 Add example authentication to test-video
12526 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12528 * gst/rtsp-server/rtsp-server.c:
12529 * gst/rtsp-server/rtsp-server.h:
12530 server: move includes back
12531 the includes are needed for sockaddr_in.
12533 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12535 * gst/rtsp-server/rtsp-client.c:
12536 * gst/rtsp-server/rtsp-client.h:
12537 * gst/rtsp-server/rtsp-server.c:
12538 * gst/rtsp-server/rtsp-server.h:
12539 rtsp: move network includes where they are needed
12541 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12543 * gst/rtsp-server/rtsp-media.h:
12544 rtsp-media.h: Minor corrections in comments.
12547 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12550 Automatic update of common submodule
12551 From e572c87 to f94d739
12553 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12557 * docs/libs/.gitignore:
12558 * examples/.gitignore:
12559 * gst/rtsp-server/.gitignore:
12562 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12564 * docs/libs/Makefile.am:
12565 docs: We don't build ps/pdf for API reference docs
12567 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12570 Automatic update of common submodule
12571 From ccbaa85 to e572c87
12573 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12576 Automatic update of common submodule
12577 From 46445ad to ccbaa85
12579 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12581 * gst/rtsp-server/Makefile.am:
12582 * gst/rtsp-server/rtsp-funnel.c:
12583 * gst/rtsp-server/rtsp-funnel.h:
12584 * gst/rtsp-server/rtsp-media.c:
12585 funnel: rename fsfunnel to rtspfunnel
12586 Rename the funnel to avoid conflicts with the farsight one.
12588 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12590 * gst/rtsp-server/Makefile.am:
12591 * gst/rtsp-server/fs-funnel.c:
12592 * gst/rtsp-server/fs-funnel.h:
12593 * gst/rtsp-server/rtsp-media.c:
12594 rtsp-media: add and use fsfunnel
12595 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12596 select-all property that we need.
12598 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12600 * gst/rtsp-server/Makefile.am:
12601 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12602 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12603 for the g-ir-compiler, rather than just assuming the env var has
12606 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12613 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12615 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12618 * gst/rtsp-server/Makefile.am:
12619 gobject-introspection: fix g-i build for uninstalled setup
12620 Requires gst-plugins-base git (> 0.10.31.2).
12622 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12624 * examples/test-uri.c:
12625 examples: add some more options and comments
12627 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12629 * gst/rtsp-server/rtsp-media-factory-uri.c:
12630 factory-uri: use right property type
12632 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12634 * gst/rtsp-server/rtsp-media-factory-uri.c:
12635 factory-uri: attempt to configure buffer-lists
12636 Attempt to configure buffer lists in the payloader for improved performance.
12638 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12640 * gst/rtsp-server/rtsp-media.c:
12641 media: attempt to configure bigger UDP buffers
12642 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12643 send buffers with high bitrate streams.
12645 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12647 * gst/rtsp-server/rtsp-client.c:
12648 client: use the socket length from getsockname
12649 Use the length returned by getsockname to perform the getnameinfo call because
12650 the size can depend on the socket type and platform.
12653 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12655 * docs/libs/gst-rtsp-server-docs.sgml:
12656 * docs/libs/gst-rtsp-server-sections.txt:
12657 docs: add uri factory to the docs
12659 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12661 * gst/rtsp-server/rtsp-client.c:
12662 * gst/rtsp-server/rtsp-media.h:
12665 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12667 * gst/rtsp-server/rtsp-client.c:
12668 * gst/rtsp-server/rtsp-media.c:
12669 * gst/rtsp-server/rtsp-media.h:
12670 * gst/rtsp-server/rtsp-session.c:
12671 * gst/rtsp-server/rtsp-session.h:
12672 rtsp-server: add support for buffer lists
12673 Add support for sending bufferlists received from appsink.
12676 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12678 * gst/rtsp-server/rtsp-client.c:
12679 * gst/rtsp-server/rtsp-media.c:
12680 * gst/rtsp-server/rtsp-media.h:
12681 * gst/rtsp-server/rtsp-sdp.c:
12682 media: make method to retrieve the play range
12683 Make a method to retrieve the playback range so that we can conditionally create
12684 a different range for the SDP and the PLAY requests.
12686 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12688 * gst/rtsp-server/rtsp-media.c:
12689 * gst/rtsp-server/rtsp-media.h:
12690 media: add signal to notify of state changes
12692 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12694 * gst/rtsp-server/rtsp-client.h:
12695 client: cleanup headers
12697 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12699 * gst/rtsp-server/rtsp-client.c:
12702 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12704 * gst/rtsp-server/rtsp-media-factory-uri.c:
12705 * gst/rtsp-server/rtsp-media-factory-uri.h:
12706 factory-uri: add support for gstpay
12707 Add an option to prefer gstpay over decoder + raw payloader.
12709 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12711 * gst/rtsp-server/rtsp-media-factory-uri.c:
12712 * gst/rtsp-server/rtsp-media-factory-uri.h:
12713 factory-uri: rework the autoplugger.
12714 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12717 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12719 * gst/rtsp-server/rtsp-media-factory-uri.c:
12720 factory-uri: use better factory filter
12721 Make better payloader filter based on autoplug rank and RTP use case.
12723 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12726 Automatic update of common submodule
12727 From 169462a to 46445ad
12729 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12731 * gst/rtsp-server/rtsp-server.c:
12732 server: set SO_REUSEADDR before bind
12733 Set the SO_REUSEADDR _before_ bind() to make it actually work.
12735 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12737 * gst/rtsp-server/rtsp-media.c:
12738 * gst/rtsp-server/rtsp-media.h:
12739 media: emit prepared signal when prepared
12740 Make a 'prepared' signal and emit it when we successfully prepared the element.
12741 This signal can be used to configure the media object after it has been prepared
12744 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
12747 Automatic update of common submodule
12748 From 011bcc8 to 169462a
12750 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
12752 python an optional dependency
12753 * configure.ac: Move up valgrind and g-i checks. Make the python
12754 dependency optional, as it was before.
12756 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12758 Merge branch 'master' into 0.11
12763 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12765 * gst/rtsp-server/rtsp-media.c:
12766 media: update range when active clients changed
12767 When we changed the number of active clients, update the current range
12768 information because we want the second client connecting to a shared resource
12769 continue from where the stream currently.
12771 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12773 * gst/rtsp-server/rtsp-media-factory-uri.c:
12774 * gst/rtsp-server/rtsp-media-factory-uri.h:
12775 factory-uri: add colorspace and fix pt
12776 Rework the way we pass data to the autoplugger.
12777 When we have raw caps, plug a converter element to make pluggin to raw
12778 payloaders more successful.
12779 Make sure all dynamically plugged payloaders have a unique payload types.
12781 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12783 * examples/Makefile.am:
12784 * examples/test-uri.c:
12785 example: add example of the uri factory
12787 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12789 * gst/rtsp-server/Makefile.am:
12790 * gst/rtsp-server/rtsp-media-factory-uri.c:
12791 * gst/rtsp-server/rtsp-media-factory-uri.h:
12792 * gst/rtsp-server/rtsp-server.h:
12793 factory-uri: add a factory to stream any URI
12794 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
12797 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12799 * gst/rtsp-server/rtsp-media.c:
12800 * gst/rtsp-server/rtsp-media.h:
12801 media: ignore spurious ASYNC_DONE messages
12802 When we are dynamically adding pads, the addition of the udpsrc elements will
12803 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
12804 the real ASYNC_DONE when everything is prerolled.
12806 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12808 * gst/rtsp-server/rtsp-media-factory.c:
12809 * gst/rtsp-server/rtsp-media-factory.h:
12810 media-factory: make lock macro
12812 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
12814 * gst/rtsp-server/rtsp-client.c:
12815 rtsp-server: Remove unused variable and dead assignment
12817 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
12819 * examples/test-launch.c:
12820 * examples/test-mp4.c:
12821 * examples/test-ogg.c:
12822 * examples/test-readme.c:
12823 * examples/test-sdp.c:
12824 * examples/test-video.c:
12825 examples: Run gst-indent
12827 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
12829 * gst/rtsp-server/rtsp-client.c:
12830 * gst/rtsp-server/rtsp-media-factory.c:
12831 * gst/rtsp-server/rtsp-media-mapping.c:
12832 * gst/rtsp-server/rtsp-media.c:
12833 * gst/rtsp-server/rtsp-params.c:
12834 * gst/rtsp-server/rtsp-sdp.c:
12835 * gst/rtsp-server/rtsp-server.c:
12836 * gst/rtsp-server/rtsp-session-pool.c:
12837 * gst/rtsp-server/rtsp-session.c:
12838 rtsp-server: Run gst-indent
12839 Since it wasn't using the upstream common previously, there was no
12840 indentation check before commiting.
12842 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
12844 * gst/rtsp-server/rtsp-media-mapping.h:
12845 * gst/rtsp-server/rtsp-media.c:
12846 * gst/rtsp-server/rtsp-media.h:
12847 * gst/rtsp-server/rtsp-sdp.c:
12848 * gst/rtsp-server/rtsp-session-pool.h:
12849 * gst/rtsp-server/rtsp-session.c:
12850 * gst/rtsp-server/rtsp-session.h:
12851 rtsp-server: Some more doc fixups
12853 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12856 Makefile: Add cruft-cleaning support
12858 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12862 * docs/Makefile.am:
12863 * docs/libs/Makefile.am:
12864 * docs/libs/gst-rtsp-server-docs.sgml:
12865 * docs/libs/gst-rtsp-server-sections.txt:
12866 * docs/libs/gst-rtsp-server.types:
12867 * docs/version.entities.in:
12868 docs: Add gtk-doc build system
12870 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12872 * gst/rtsp-server/Makefile.am:
12873 Makefile.am: Use standard GIR make behaviour
12875 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12879 autogen/configure: Bring more in sync to standard gst module behaviour
12881 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12883 * gst/rtsp-server/rtsp-media.c:
12884 media: warn and fail when gstrtpbin is not found
12886 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12889 configure: open 0.11 branch
12891 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
12895 Add common submodule
12897 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
12899 * common/ChangeLog:
12900 * common/Makefile.am:
12901 * common/c-to-xml.py:
12902 * common/check.mak:
12903 * common/coverage/coverage-report-entry.pl:
12904 * common/coverage/coverage-report.pl:
12905 * common/coverage/coverage-report.xsl:
12906 * common/coverage/lcov.mak:
12907 * common/gettext.patch:
12908 * common/glib-gen.mak:
12909 * common/gst-autogen.sh:
12910 * common/gst-xmlinspect.py:
12912 * common/gstdoc-scangobj:
12913 * common/gtk-doc-plugins.mak:
12914 * common/gtk-doc.mak:
12915 * common/m4/.gitignore:
12916 * common/m4/Makefile.am:
12917 * common/m4/README:
12918 * common/m4/as-ac-expand.m4:
12919 * common/m4/as-auto-alt.m4:
12920 * common/m4/as-compiler-flag.m4:
12921 * common/m4/as-compiler.m4:
12922 * common/m4/as-docbook.m4:
12923 * common/m4/as-libtool-tags.m4:
12924 * common/m4/as-libtool.m4:
12925 * common/m4/as-python.m4:
12926 * common/m4/as-scrub-include.m4:
12927 * common/m4/as-version.m4:
12928 * common/m4/ax_create_stdint_h.m4:
12929 * common/m4/check.m4:
12930 * common/m4/glib-gettext.m4:
12931 * common/m4/gst-arch.m4:
12932 * common/m4/gst-args.m4:
12933 * common/m4/gst-check.m4:
12934 * common/m4/gst-debuginfo.m4:
12935 * common/m4/gst-default.m4:
12936 * common/m4/gst-doc.m4:
12937 * common/m4/gst-error.m4:
12938 * common/m4/gst-feature.m4:
12939 * common/m4/gst-function.m4:
12940 * common/m4/gst-gettext.m4:
12941 * common/m4/gst-glib2.m4:
12942 * common/m4/gst-libxml2.m4:
12943 * common/m4/gst-plugindir.m4:
12944 * common/m4/gst-valgrind.m4:
12945 * common/m4/gtk-doc.m4:
12946 * common/m4/introspection.m4:
12947 * common/m4/pkg.m4:
12948 * common/mangle-tmpl.py:
12949 * common/plugins.xsl:
12951 * common/release.mak:
12952 * common/scangobj-merge.py:
12953 * common/upload.mak:
12954 common: Remove static version
12956 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
12958 * common/m4/introspection.m4:
12959 Update introspection.m4 to match usage
12961 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12965 Remove old stuff from the README
12967 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12970 back to development
12972 === release 0.10.7 ===
12974 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12979 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12981 * examples/test-ogg.c:
12982 test-ogg: remove parsers
12983 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
12984 buffers with timestamps. Using the parsers also seems to break things.
12986 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12988 * bindings/vala/gst-rtsp-server-0.10.vapi:
12989 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12990 Updated Vala bindings
12992 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12994 * common/m4/introspection.m4:
12996 * gst/rtsp-server/Makefile.am:
12997 Added initial gobject-introspection support
12999 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13001 * gst/rtsp-server/rtsp-media-factory.c:
13002 media-factory: don't use host for shared hash key
13003 When we generate the key to share made between connections, don't include the
13004 host used to connect so that we can share media even if between clients that
13005 connected with localhost and ones with the ip address.
13007 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13009 * bindings/vala/Makefile.am:
13010 build: fix distcheck
13012 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13014 * bindings/vala/gst-rtsp-server-0.10.vapi:
13015 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13016 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13017 Update Vala bindings
13019 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13021 * bindings/vala/Makefile.am:
13023 Fix configure checks and installation location for Vala bindings
13026 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13029 back to development
13031 === release 0.10.6 ===
13033 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13036 configure: release 0.10.6
13038 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13040 * gst/rtsp-server/rtsp-media.c:
13041 media: help the compiler a little
13043 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13045 * gst/rtsp-server/rtsp-media.c:
13046 * gst/rtsp-server/rtsp-media.h:
13047 * gst/rtsp-server/rtsp-session.c:
13048 media: cleanup media transport before freeing
13049 Cleanup the media transport data before freeing. In particular, remove the qdata
13050 from the rtpsource object.
13052 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13054 * gst/rtsp-server/rtsp-media-factory.c:
13055 * gst/rtsp-server/rtsp-media-factory.h:
13056 * gst/rtsp-server/rtsp-media.c:
13057 * gst/rtsp-server/rtsp-media.h:
13058 media-factory: add eos-shutdown property
13059 Add an eos-shutdown property that will send an EOS to the pipeline before
13060 shutting it down. This allows for nice cleanup in case of a muxer.
13063 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13065 * gst/rtsp-server/rtsp-media.c:
13066 * gst/rtsp-server/rtsp-media.h:
13067 media: use multiudpsink send-duplicates when we can
13068 If we have a new enough multiudpsink with the send-duplicates property, use this
13069 instead of doing our own filtering. Our custom filtering code should eventually
13070 be removed when we can depend on a released -good.
13072 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13074 * gst/rtsp-server/rtsp-media.c:
13075 media: don't leak destinations
13076 Refactor and cleanup the destinations array when the stream is destroyed.
13078 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13080 * gst/rtsp-server/rtsp-media.c:
13081 * gst/rtsp-server/rtsp-media.h:
13082 media: don't add udp addresses multiple times
13083 Keep track of the udp addresses we added to udpsink and never add the same udp
13084 destination twice. This avoids duplicate packets when using multicast.
13086 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13088 * gst/rtsp-server/rtsp-server.c:
13089 server: disable use of SO_LINGER
13090 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13091 server close()s the connection.
13093 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13095 * gst/rtsp-server/rtsp-server.c:
13096 server: use 5 second linger period in SO_LINGER
13097 Wait 5 seconds before clearing the send buffers and reseting the connection with
13098 the client when we do a close. This should be enough time to get the message to
13102 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13104 * gst/rtsp-server/rtsp-server.c:
13105 server: use SO_LINGER
13106 SO_LINGER on the socket will make sure that any pending data on the socket is
13107 flushed ASAP and that the socket connection is reset. This makes sure that the
13108 socket can be reused immediately.
13111 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13114 README: add blurb about shared media factories
13116 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13118 * gst/rtsp-server/rtsp-media.c:
13119 Add stdlib.h for atoi()
13121 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13123 * bindings/python/Makefile.am:
13124 * bindings/vala/Makefile.am:
13125 build: distcheck fixes
13126 Fix 'make distcheck', somewhat (it still fails because it tries to
13127 install files into /usr/share/vala/vapi/ irrespective of the
13128 configured prefix).
13130 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13133 configure: bump core/base requirements to released version
13134 Makes things less confusing for people.
13136 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13139 configure: fail if GStreamer core/base requirements are not met
13141 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13143 * gst/rtsp-server/rtsp-client.c:
13144 client: improve client cleanups
13145 Make sure the session does not timeout when using TCP. We need to do this
13146 because quicktime player does not send RTCP for some reason in tunneled
13148 Refactor some cleanup code.
13151 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13153 * gst/rtsp-server/rtsp-session.c:
13154 * gst/rtsp-server/rtsp-session.h:
13155 session: add support for prevent session timeouts
13156 Add an atomix counter to prevent session timeouts when we are, for example,
13157 streaming over TCP.
13159 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13161 * gst/rtsp-server/rtsp-client.c:
13162 client: fix unlink on session timeouts
13163 When our session times out, make sure we unlink all streams in this
13165 Remove the tunnelid when closing the connection.
13167 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13169 * gst/rtsp-server/rtsp-session.c:
13170 session: small cleanups
13172 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13174 * gst/rtsp-server/rtsp-client.c:
13175 client: handle lost_tunnel callbacks
13176 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13177 hashtable so that we can reuse it for when the client reopens the POST
13179 Close the connection after a TEARDOWN.
13180 Make sure or watchid is cleared when the watch is removed.
13183 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13185 * gst/rtsp-server/rtsp-client.c:
13186 * gst/rtsp-server/rtsp-media.c:
13187 * gst/rtsp-server/rtsp-sdp.c:
13188 rtsp-server: add more support for multicast
13190 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13193 * gst/rtsp-server/rtsp-media.c:
13194 * gst/rtsp-server/rtsp-media.h:
13195 media: allow configuration of allowed lower transport
13197 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13199 * gst/rtsp-server/rtsp-client.h:
13200 * gst/rtsp-server/rtsp-media.c:
13201 * gst/rtsp-server/rtsp-media.h:
13202 * gst/rtsp-server/rtsp-sdp.c:
13203 * gst/rtsp-server/rtsp-sdp.h:
13204 * gst/rtsp-server/rtsp-server.c:
13205 rtsp: keep track of server ip and ipv6
13206 Keep track of how the client connected to the server and setup the udp ports
13207 with the same protocol.
13208 Copy the server ip address in the SDP so that clients can send RTCP back to
13211 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13213 * gst/rtsp-server/rtsp-session.c:
13216 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13218 * gst/rtsp-server/rtsp-client.c:
13219 client: use right size for malloc
13221 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13223 * gst/rtsp-server/rtsp-server.c:
13224 server: comment ipv6 server listening address
13226 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13228 * gst/rtsp-server/rtsp-media.c:
13229 media: allow for ipv6 sockets
13231 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13233 * gst/rtsp-server/rtsp-server.c:
13234 * gst/rtsp-server/rtsp-server.h:
13235 server: rework server part
13236 Allow setting a bind address, make sure we can deal with ipv6.
13237 Remove the port property and change with the service property.
13239 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13241 * gst/rtsp-server/rtsp-media.h:
13242 media: update comments a little
13244 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13246 * gst/rtsp-server/rtsp-client.c:
13247 client: make content-base better
13248 Use the URI formatting functions to make a content-base. Also make sure that
13249 there is a trailing / at the end.
13251 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13253 * gst/rtsp-server/rtsp-client.c:
13254 client: guard against invalid paths
13256 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13258 * examples/test-video.c:
13259 test: catch server bind errors
13261 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13263 * gst/rtsp-server/rtsp-media.c:
13264 rtspmedia: emit "unprepared" if _prepare fails.
13265 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13266 media object is removed from its factory's cache.
13268 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13270 * gst/rtsp-server/rtsp-media.c:
13271 media: collect media position when seek completes
13273 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13275 * gst/rtsp-server/rtsp-client.c:
13276 client: call unlink_streams in client finalize
13279 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13281 * gst/rtsp-server/rtsp-media.c:
13282 media: limit the time to wait to something huge
13283 Avoid waiting forever but limit the timeout to 20 seconds.
13285 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13287 * gst/rtsp-server/rtsp-sdp.c:
13288 sdp: reindent and check for prepared status
13290 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13292 * gst/rtsp-server/rtsp-media.c:
13293 * gst/rtsp-server/rtsp-media.h:
13294 * gst/rtsp-server/rtsp-session.c:
13295 media: avoid doing _get_state() for state changes
13296 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13297 until the media is prerolled or in error. This avoids doing a blocking call of
13298 gst_element_get_state() that can cause lockups when there is an error.
13301 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13303 * gst/rtsp-server/rtsp-media.c:
13306 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13308 * gst/rtsp-server/rtsp-media-factory.c:
13309 media-factory: better error handling
13310 Improve the error handling a bit.
13312 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13314 * gst/rtsp-server/rtsp-client.c:
13315 client: rework transport parsing
13316 Rework the transport parsing code so that we can ignore transports we don't
13317 support instead of just picking the first one we can parse.
13318 Configure a (for now hardcoded) destination for multicast transports.
13320 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13322 * gst/rtsp-server/rtsp-media.c:
13323 media: set multicast sink parameters
13324 Disable loop and automatic multicast join on the udpsink elements.
13325 Add some more debug info.
13326 Reset some state variables in the right place.
13327 Use the right port numbers for multicast.
13329 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13331 * gst/rtsp-server/rtsp-session.c:
13332 session: handle transport setup correctly
13333 Handle UDP, MCAST and TCP transport negotiation more correctly.
13334 Store the server session SSRC in the transport.
13336 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13338 * gst/rtsp-server/rtsp-client.c:
13339 rtsp-client: implement error_full
13340 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13343 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13346 * gst/rtsp-server/rtsp-client.c:
13347 * gst/rtsp-server/rtsp-server.c:
13348 docs: update docs and comments
13350 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13352 * gst/rtsp-server/rtsp-sdp.c:
13353 sdp: make server work better when behind a proxy
13355 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13357 * gst/rtsp-server/rtsp-client.c:
13358 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13360 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13362 * gst/rtsp-server/rtsp-client.c:
13363 * gst/rtsp-server/rtsp-media-factory.c:
13364 * gst/rtsp-server/rtsp-media-mapping.c:
13365 * gst/rtsp-server/rtsp-media.c:
13366 * gst/rtsp-server/rtsp-server.c:
13367 * gst/rtsp-server/rtsp-session-pool.c:
13368 * gst/rtsp-server/rtsp-session.c:
13369 Use GStreamer's debugging subsystem
13371 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13373 * gst/rtsp-server/rtsp-media-factory.c:
13374 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13376 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13379 back to development
13381 === release 0.10.5 ===
13383 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13388 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13391 configure: bump required versions
13393 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13395 * gst/rtsp-server/rtsp-client.c:
13396 client: call weak-unref on client->sessions from finalize
13399 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13401 * gst/rtsp-server/rtsp-media.c:
13402 media: Fixed crasher where caps got unref'ed too often
13404 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13407 * pkgconfig/.gitignore:
13408 * pkgconfig/Makefile.am:
13409 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13410 Added pkg-config file to use gst-rtsp-server uninstalled
13412 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13414 * gst/rtsp-server/rtsp-media.c:
13415 media: add some docs
13417 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13419 * gst/rtsp-server/rtsp-client.c:
13420 rtsp: Use gst_rtsp_watch_send_message().
13421 Use gst_rtsp_watch_send_message() since the old API which used
13422 gst_rtsp_watch_queue_message() has been deprecated.
13424 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13427 back to development
13429 === release 0.10.4 ===
13431 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13436 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13438 * gst/rtsp-server/rtsp-client.c:
13439 * gst/rtsp-server/rtsp-session.c:
13440 * gst/rtsp-server/rtsp-session.h:
13441 rtsp: allocate channels in TCP mode
13442 When the client does not provide us with channels in TCP mode, allocate channels
13445 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13447 * gst/rtsp-server/rtsp-client.c:
13448 client: don't crash when tunnelid is missing
13449 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13450 don't crash but return an error response to the client.
13453 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13455 * bindings/vala/gst-rtsp-server-0.10.vapi:
13456 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13457 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13458 bindings: update vala bindings with new method
13460 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13462 * gst/rtsp-server/rtsp-session-pool.c:
13463 * gst/rtsp-server/rtsp-session-pool.h:
13464 sessionpool: add function to filter sessions
13465 Add generic function to retrieve/remove sessions.
13467 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13470 configure: bump core/base requirements to release
13472 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13474 * gst/rtsp-server/rtsp-media.c:
13475 media: fix indentation
13477 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13479 * gst/rtsp-server/rtsp-media.c:
13480 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13482 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13484 * gst/rtsp-server/rtsp-media.c:
13485 set state and remove elements of media in for loop
13487 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13489 * bindings/vala/gst-rtsp-server-0.10.vapi:
13490 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13491 Added gst_rtsp_media_remove_elements function to Vala bindings
13493 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13495 * gst/rtsp-server/rtsp-media.c:
13496 * gst/rtsp-server/rtsp-media.h:
13497 Added gst_rtsp_media_remove_elements function
13499 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13501 * gst/rtsp-server/rtsp-media.c:
13502 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13504 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13506 * bindings/vala/gst-rtsp-server-0.10.vapi:
13507 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13508 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13509 Updated Vala bindings
13511 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13513 * gst/rtsp-server/rtsp-media.c:
13514 * gst/rtsp-server/rtsp-media.h:
13515 Added vmethod unprepare to GstRTSPMedia
13516 The default implementation sets the state of the pipeline to GST_STATE_NULL
13518 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13520 * gst/rtsp-server/rtsp-media-factory.c:
13521 * gst/rtsp-server/rtsp-media-factory.h:
13522 Made collect_streams function public
13524 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13526 * gst/rtsp-server/rtsp-media-factory.c:
13527 * gst/rtsp-server/rtsp-media-factory.h:
13528 * gst/rtsp-server/rtsp-media.c:
13529 Added vmethod create_pipeline to GstRTSPMediaFactory
13530 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13532 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13534 * gst/rtsp-server/rtsp-client.c:
13535 client: use g_source_destroy()
13536 We need to use g_source_destroy() because we might have added the source to a
13537 different main context than the default one.
13539 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13541 * gst/rtsp-server/Makefile.am:
13542 * gst/rtsp-server/rtsp-client.c:
13543 * gst/rtsp-server/rtsp-params.c:
13544 * gst/rtsp-server/rtsp-params.h:
13545 rtsp: prepare for handling GET/SET_PARAMETER
13546 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13548 Fix return codes of handlers.
13550 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13552 * gst/rtsp-server/rtsp-media.c:
13553 media: don't leak session pads
13555 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13557 * gst/rtsp-server/rtsp-media.c:
13558 media: clean up the messages a bit
13560 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13562 * gst/rtsp-server/rtsp-sdp.c:
13563 sdp: warn and skip streams without media
13565 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13567 * bindings/vala/gst-rtsp-server-0.10.vapi:
13568 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13569 vala: Fixed typo in header file of RTSPMediaStream
13571 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13573 * gst/rtsp-server/rtsp-media.c:
13575 Fix a debug message
13576 Make dumping RTCP stats configurable
13578 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13580 * gst/rtsp-server/rtsp-media.c:
13581 media: be less verbose and leak less
13583 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13585 * gst/rtsp-server/rtsp-media.c:
13586 media: don't leak the destination address
13588 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13590 * gst/rtsp-server/rtsp-client.c:
13591 * gst/rtsp-server/rtsp-media.c:
13592 * gst/rtsp-server/rtsp-media.h:
13593 * gst/rtsp-server/rtsp-session.c:
13594 * gst/rtsp-server/rtsp-session.h:
13595 rtsp: use RTCP to keep the session alive
13596 Use the RTCP rtcp-from stats field to find the associated session and use this
13597 to keep the session alive.
13599 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13601 * gst/rtsp-server/rtsp-session.c:
13602 session: add 5sec to the real session timeout
13603 Allow the session to live 5sec longer before really timing out. This should give
13604 clients some extra time to keep the session active.
13606 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13608 * gst/rtsp-server/rtsp-client.c:
13609 client: replay OK to GET/SET_PARAMETER
13610 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13611 so that we return OK for those requests.
13613 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13615 * gst/rtsp-server/rtsp-media.c:
13616 * gst/rtsp-server/rtsp-media.h:
13617 media: keep track of active transports
13618 Keep track of which transport is active to avoid closing the connection too
13620 Remove the destination transport also when going to NULL.
13621 Print some stats about the SDES and other RTCP messages we receive from the
13624 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13626 * examples/.gitignore:
13627 * examples/Makefile.am:
13628 * examples/test-sdp.c:
13629 example: add SDP relay example
13631 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13633 * gst/rtsp-server/rtsp-media.c:
13634 media: also count active TCP connections
13636 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13638 * gst/rtsp-server/rtsp-media-factory.c:
13639 * gst/rtsp-server/rtsp-media.c:
13640 * gst/rtsp-server/rtsp-media.h:
13641 rtsp: add support for dynamic elements
13642 Add support for dynamic elements.
13643 Don't set live pipelines back to paused.
13645 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13647 * gst/rtsp-server/rtsp-sdp.c:
13648 sdp: don't add encoding name when absent in caps
13650 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13652 * gst/rtsp-server/rtsp-client.c:
13653 client: warn when we can't do RTP-Info
13655 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13657 * gst/rtsp-server/rtsp-media-factory.c:
13658 factory: factor out the stream construction
13660 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13662 * gst/rtsp-server/rtsp-client.c:
13663 client: only add RTP-Info when we have the info
13664 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13667 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13670 back to development
13672 === release 0.10.3 ===
13674 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13678 - Fixes a bug where it put the wrong verion in pkgconfig
13679 - Link RTP and RTCP sources
13681 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13683 * gst/rtsp-server/rtsp-media.c:
13684 * gst/rtsp-server/rtsp-media.h:
13685 media: link the RTP udpsrc to the session manager
13686 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13687 shut down when the client sends a packet to open firewalls.
13689 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13691 * pkgconfig/gst-rtsp-server.pc.in:
13692 Don't use hard-coded version number in pkg-config file
13694 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13697 back to development
13699 === release 0.10.2 ===
13701 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13706 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13709 * common/m4/.gitignore:
13710 * examples/.gitignore:
13711 * pkgconfig/.gitignore:
13712 add some .gitignore files
13714 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13716 * gst/rtsp-server/rtsp-media.c:
13717 media: seek to key frames
13719 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13721 * gst/rtsp-server/rtsp-media.c:
13722 media: emit the unprepared signal by id
13723 Emit the unprepared signal by id instead of name and set the media as
13726 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13728 * gst/rtsp-server/rtsp-media.c:
13729 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13731 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13733 * gst/rtsp-server/rtsp-server.c:
13734 Added finalize function to GstRTPSPServer to unref session pool and media mapping
13736 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13738 * bindings/vala/gst-rtsp-server-0.10.vapi:
13739 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13740 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13741 Updated vala bindings
13743 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13745 * gst/rtsp-server/Makefile.am:
13746 * gst/rtsp-server/rtsp-client.c:
13747 * gst/rtsp-server/rtsp-media.c:
13748 server: use appsink and appsrc with the API
13749 Use the appsink/appsrc API instead of the signals for higher
13752 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13754 * examples/test-ogg.c:
13755 tests: set the payload type correctly
13757 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13759 * gst/rtsp-server/rtsp-media-factory.c:
13760 factory: connect to the unprepare signal
13761 Connect to the unprepare signal for non-reusable media so that we can remove
13762 them from the cache.
13764 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13766 * gst/rtsp-server/rtsp-media.c:
13767 * gst/rtsp-server/rtsp-media.h:
13768 media: add signal to notify of unprepare
13770 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13772 * gst/rtsp-server/rtsp-media.c:
13773 * gst/rtsp-server/rtsp-media.h:
13774 media: more work on making the media shared
13775 Add a reusable flag to medias, indicating that they can be reused after a state
13779 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13781 * examples/test-readme.c:
13782 examples: mark the example as shared for testing
13784 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13786 * gst/rtsp-server/rtsp-media.c:
13787 * gst/rtsp-server/rtsp-media.h:
13788 client: support shared media
13789 Always perform the state actions even if the target state of the pipeline is
13790 already correct, we still want to add/remove the transports when we are dealing
13792 Keep a counter of the number of active transports for a media so that we can use
13793 this to perform a state change when needed.
13794 Perform a state change of the pipeline only when the first transport was added
13795 or when there are no active transports.
13797 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13799 * gst/rtsp-server/rtsp-client.c:
13800 client: fix refcounting crasher
13801 Don't need to remove the weak refs in the finalize methods, they are already
13802 removed in the dispose.
13803 Don't register the callback with a DestroyNofity.
13805 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13807 * gst/rtsp-server/rtsp-client.c:
13808 Fix rtsp client refcount management in TCP mode.
13809 Don't unref a client ref we never had. Fixes an unref
13810 of an already-free client object after a client
13811 teardown request for me.
13813 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13815 * gst/rtsp-server/rtsp-session.c:
13816 docs: fix typo in API docs
13818 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13820 * gst/rtsp-server/rtsp-media.c:
13821 More seeking fixes.
13822 Keep the udp sources in playing even if we go to paused. unlock the sources when
13824 Add some more debug info.
13825 Only seek when we need to.
13826 Keep track of the position when we go to paused.
13828 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13830 * gst/rtsp-server/rtsp-client.c:
13831 * gst/rtsp-server/rtsp-media.c:
13832 * gst/rtsp-server/rtsp-media.h:
13833 Add beginnings of seeking.
13834 Parse the Range header and perform a seek on the pipeline for the requested
13835 position. It's disabled currently until I figure out what's going wrong.
13837 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13839 * gst/rtsp-server/rtsp-client.c:
13840 allow pause requests for now.
13843 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13845 * gst/rtsp-server/rtsp-client.c:
13846 Remove weak ref on the session in teardown
13847 We need to remove our weakref from the session when we do a teardown because
13848 else we close the TCP connection prematurely.
13850 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13852 * gst/rtsp-server/rtsp-client.c:
13853 * gst/rtsp-server/rtsp-client.h:
13854 * gst/rtsp-server/rtsp-session-pool.c:
13855 Do some more session cleanup
13856 Make session timeout kill the TCP connection that currently watches the
13858 Remove the client timeout property.
13860 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13862 * gst/rtsp-server/rtsp-client.c:
13863 * gst/rtsp-server/rtsp-client.h:
13864 * gst/rtsp-server/rtsp-media.c:
13865 * gst/rtsp-server/rtsp-media.h:
13866 * gst/rtsp-server/rtsp-server.c:
13867 * gst/rtsp-server/rtsp-session.c:
13868 * gst/rtsp-server/rtsp-session.h:
13870 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
13873 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13875 * examples/Makefile.am:
13876 * examples/test-launch.c:
13877 Add example server that takes launch lines
13878 Add an example server that streams any -launch line.
13880 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13882 * examples/test-readme.c:
13883 * gst/rtsp-server/rtsp-client.c:
13884 * gst/rtsp-server/rtsp-media.c:
13885 * gst/rtsp-server/rtsp-media.h:
13886 Add support for live streams
13887 Add support for live streams and ranges
13888 Start on handling TCP data transfer.
13890 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13892 * gst/rtsp-server/rtsp-media.c:
13893 Free the pipeline before other things
13896 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13898 * gst/rtsp-server/rtsp-client.c:
13899 Only free the pending tunnel if there is one
13902 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13904 * gst/rtsp-server/rtsp-client.c:
13905 * gst/rtsp-server/rtsp-client.h:
13906 * gst/rtsp-server/rtsp-media.c:
13907 rtsp-server: Add support for tunneling
13908 Add support for tunneling over HTTP.
13909 Use new connection methods to retrieve the url.
13910 Dispatch messages based on the message type instead of blindly
13911 assuming it's always a request.
13912 Keep track of the watch id so that we can remove it later.
13913 Set the media pipeline to NULL before unreffing the pipeline.
13915 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13917 * gst/rtsp-server/rtsp-client.c:
13918 * gst/rtsp-server/rtsp-client.h:
13919 Fix for channel -> watch rename in gstreamer
13920 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
13922 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13924 * gst/rtsp-server/rtsp-client.c:
13925 * gst/rtsp-server/rtsp-client.h:
13927 Use the async RTSP channels instead of spawning a new thread for each client.
13928 If a sessionid is specified in a request, fail if we don't have the session.
13930 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13932 * gst/rtsp-server/rtsp-media.c:
13933 Add better debug info
13934 Add some better debug info.
13936 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13938 * examples/test-video.c:
13940 Add support for session timeouts in the example.
13942 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13944 * gst/rtsp-server/rtsp-session-pool.c:
13945 * gst/rtsp-server/rtsp-session-pool.h:
13946 Pass GTimeVal around for performance reasons
13947 Get the current time only once and pass it around so that sessions don't have to
13948 get the current time anymore.
13949 Add experimental support for a GSource that dispatches when the session needs to
13952 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13954 * gst/rtsp-server/rtsp-session.c:
13955 * gst/rtsp-server/rtsp-session.h:
13956 Add better support for session timeouts
13957 Add a method to request the number of milliseconds when a session will timeout.
13959 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13961 * gst/rtsp-server/rtsp-media.c:
13962 * gst/rtsp-server/rtsp-media.h:
13963 Add suport for RTP manager monitoring
13964 Add the first stage in monitoring the rtp manager.
13965 Make sure we don't update the state to something we don't want.
13967 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13969 * gst/rtsp-server/rtsp-client.c:
13970 Add support for session keepalive
13971 Get and update the session timeout for all requests. get the session as early as
13974 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13976 * gst/rtsp-server/rtsp-media-factory.h:
13977 * gst/rtsp-server/rtsp-media.c:
13978 * gst/rtsp-server/rtsp-media.h:
13979 Handle media bus messages
13980 Handle media bus messages in a custom mainloop and dispatch them to the
13981 RTSPMedia objects. Let the default implementation handle some common messages.
13983 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13985 * gst/rtsp-server/rtsp-client.c:
13986 * gst/rtsp-server/rtsp-session-pool.c:
13987 * gst/rtsp-server/rtsp-session.c:
13988 Some more session timeout handling
13989 Move the session header setting code to a central place so that we always add
13990 the timeout parameter too.
13991 Handle timeouts by running the session cleanup code.
13992 Stop media before cleaning up.
13994 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13996 * gst/rtsp-server/rtsp-client.c:
13997 * gst/rtsp-server/rtsp-client.h:
13998 Add timeout property
13999 Add a timeout property ot the client and make the other properties into GObject
14002 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14004 * gst/rtsp-server/rtsp-session-pool.c:
14005 Use getters and setters in property code
14006 Use the getters and setters for the timeout property instead of locking
14009 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14011 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
14013 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14015 * gst/rtsp-server/rtsp-session-pool.c:
14016 * gst/rtsp-server/rtsp-session-pool.h:
14017 * gst/rtsp-server/rtsp-session.c:
14018 * gst/rtsp-server/rtsp-session.h:
14019 Add more timeout stuff
14020 Add method to check if a session is expired.
14021 Add method to perform cleanup on a session pool.
14023 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14025 * gst/rtsp-server/rtsp-client.c:
14026 * gst/rtsp-server/rtsp-session-pool.c:
14027 * gst/rtsp-server/rtsp-session-pool.h:
14028 * gst/rtsp-server/rtsp-session.c:
14029 * gst/rtsp-server/rtsp-session.h:
14030 Add beginnings of session timeouts and limits
14031 Add the timeout value to the Session header for unusual timeout values.
14032 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14033 limit on the amount of retry we do after a sessionid collision.
14034 Add properties to the sessionid and the timeout of a session. Keep track of
14035 creation time and last access time for sessions.
14037 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14039 * gst/rtsp-server/rtsp-client.c:
14040 * gst/rtsp-server/rtsp-media.c:
14041 * gst/rtsp-server/rtsp-media.h:
14042 * gst/rtsp-server/rtsp-sdp.c:
14043 * gst/rtsp-server/rtsp-session-pool.c:
14044 * gst/rtsp-server/rtsp-session.c:
14045 * gst/rtsp-server/rtsp-session.h:
14046 Cleanup of sessions and more
14047 Fix the refcounting of media and sessions in the client. Properly clean up the
14048 session data when the client performs a teardown.
14049 Add Server header to responses.
14050 Allow for multiple uri setups in one session.
14051 Add Range header to the PLAY response and add the range attribute to the SDP
14053 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14054 give the ownership of the sessionid to the session object.
14056 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14058 * gst/rtsp-server/rtsp-server.c:
14059 * gst/rtsp-server/rtsp-server.h:
14061 Rename the 'server_port' variable to simply 'port'.
14063 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14066 * gst/rtsp-server/rtsp-client.c:
14067 * gst/rtsp-server/rtsp-media.c:
14068 * gst/rtsp-server/rtsp-media.h:
14069 * gst/rtsp-server/rtsp-session.c:
14070 * gst/rtsp-server/rtsp-session.h:
14071 Rework the way we handle transports for streams
14072 Make the media accept an array of transports for the streams that we have
14073 configured for the play/pause requests.
14074 Implement server states for a client and its media.
14075 Require 0.10.22.1 (git HEAD) of gstreamer.
14077 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14079 * gst/rtsp-server/rtsp-client.c:
14080 * gst/rtsp-server/rtsp-media-factory.c:
14081 Drop const from functions dealing with urls
14082 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14083 have the right const in them.
14085 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14087 * gst/rtsp-server/rtsp-client.c:
14088 * gst/rtsp-server/rtsp-media.c:
14089 * gst/rtsp-server/rtsp-sdp.c:
14093 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14095 * gst/rtsp-server/rtsp-client.c:
14096 * gst/rtsp-server/rtsp-media-factory.c:
14097 * gst/rtsp-server/rtsp-media.c:
14098 * gst/rtsp-server/rtsp-media.h:
14100 Don't keep a reference to the GstRTSPMedia in the stream.
14101 Free more things when freeing the GstRTSPMedia.
14103 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14106 * gst/rtsp-server/rtsp-media-factory.c:
14107 * gst/rtsp-server/rtsp-media-factory.h:
14108 * gst/rtsp-server/rtsp-media.c:
14109 * gst/rtsp-server/rtsp-media.h:
14110 * gst/rtsp-server/rtsp-server.c:
14111 * gst/rtsp-server/rtsp-server.h:
14112 More docs and small cleanups
14113 Add some more docs and update the README
14114 Cleanup some method names.
14115 Remove an unneeded idx field in the GstRTSPMediaStream
14117 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14120 * examples/Makefile.am:
14121 * examples/test-readme.c:
14122 Add a README and more example code
14123 Add a README file that contains a small introduction on how to use the server
14124 along with the example code explained in the readme.
14126 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14128 * gst/rtsp-server/rtsp-media.c:
14129 * gst/rtsp-server/rtsp-server.c:
14130 Fix some leaks and change default port
14131 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14132 we finished the initial preroll. If we keep them locked, setting the pipeline to
14133 NULL will not stop and clean up the sources correctly.
14134 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14136 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14138 * gst/rtsp-server/rtsp-session.c:
14139 * gst/rtsp-server/rtsp-session.h:
14140 Cleanups to the session object
14141 Remove some unneeded variables in the session state of a stream such as the
14142 owner media and the server transport.
14143 Get the configuration of a media stream in a session based on the media_stream
14144 in the original object instead of our cached index.
14145 Free more data in the finalize method.
14147 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14149 * gst/rtsp-server/rtsp-client.c:
14150 * gst/rtsp-server/rtsp-client.h:
14151 Cleanups and reuse media from DESCRIBE
14152 Handle thread create errors.
14153 Rename some internal methods to better match what they actually do.
14154 Handle misconfiguration of session_pool and media_mapping gracefully.
14155 Cache the DESCRIBE media and uri in the client connection and reuse them when
14156 we receive a SETUP request in the same connection for the same uri.
14157 Cleanup the client connection object.
14159 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14161 * gst/rtsp-server/rtsp-media-factory.c:
14162 * gst/rtsp-server/rtsp-media-factory.h:
14163 * gst/rtsp-server/rtsp-media.c:
14164 * gst/rtsp-server/rtsp-media.h:
14165 Add shared properties to media and factory
14166 Add the shared property to media.
14167 Implement some simple caching in the factory depending on if the media is shared
14170 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14172 * gst/rtsp-server/rtsp-client.c:
14173 Add a little comment
14174 Add some comment about the content-base header.
14176 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14178 * examples/Makefile.am:
14179 * examples/test-mp4.c:
14180 * examples/test-ogg.c:
14181 * examples/test-video.c:
14182 * gst/rtsp-server/Makefile.am:
14183 * gst/rtsp-server/rtsp-client.c:
14184 * gst/rtsp-server/rtsp-client.h:
14185 * gst/rtsp-server/rtsp-media-factory.c:
14186 * gst/rtsp-server/rtsp-media-factory.h:
14187 * gst/rtsp-server/rtsp-media.c:
14188 * gst/rtsp-server/rtsp-media.h:
14189 * gst/rtsp-server/rtsp-sdp.c:
14190 * gst/rtsp-server/rtsp-sdp.h:
14191 * gst/rtsp-server/rtsp-server.c:
14192 * gst/rtsp-server/rtsp-server.h:
14193 * gst/rtsp-server/rtsp-session.c:
14194 * gst/rtsp-server/rtsp-session.h:
14195 Reorganize things, prepare for media sharing
14196 Added various other test server examples
14197 Move the SDP message generation to a separate helper.
14198 Refactor common code for finding the session.
14199 Add content-base for realplayer compatibility
14200 Clean up request uris before processing for better vlc compatibility.
14201 Move prerolling and pipeline construction to the RTSPMedia object.
14202 Use multiudpsink for future pipeline reuse.
14204 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14207 Back to development
14210 === release 0.10.1 ===
14212 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14215 Make 0.10.1 release
14218 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14220 * bindings/vala/Makefile.am:
14222 Add more directories and files to the dist.
14224 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14226 * bindings/python/Makefile.am:
14227 * bindings/python/rtspserver.override:
14228 Fixed compile error of python bindings
14230 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14232 * bindings/vala/gst-rtsp-server-0.10.vapi:
14233 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14234 Marked values as nullable accordingly
14236 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14238 * bindings/vala/gst-rtsp-server-0.10.vapi:
14239 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14240 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14241 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14242 Updated Vala bindings
14244 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14246 * gst/rtsp-server/rtsp-client.c:
14247 * gst/rtsp-server/rtsp-media-mapping.c:
14248 * gst/rtsp-server/rtsp-media-mapping.h:
14249 * gst/rtsp-server/rtsp-media.h:
14250 * gst/rtsp-server/rtsp-session-pool.h:
14251 Cleanups and doc updates
14252 Add some more documentation and do some minor cleanups here and there.
14254 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14256 * gst/rtsp-server/rtsp-client.c:
14257 * gst/rtsp-server/rtsp-media-factory.c:
14258 * gst/rtsp-server/rtsp-media-factory.h:
14259 * gst/rtsp-server/rtsp-media.c:
14260 * gst/rtsp-server/rtsp-media.h:
14261 * gst/rtsp-server/rtsp-session.c:
14262 * gst/rtsp-server/rtsp-session.h:
14264 Rename GstRTSPMediaBin to GstRTSPMedia
14265 Parse the request url into a GstRTSPUri object and pass this object to the
14266 various handlers and methods that require the uri.
14268 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14272 Add some more docs and remove some old code from the example.
14274 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14276 * gst/rtsp-server/rtsp-client.c:
14277 Handle state change failures better
14278 Handle state change failures better when changing the state of the pipeline to
14281 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14283 * gst/rtsp-server/rtsp-media-factory.c:
14284 * gst/rtsp-server/rtsp-media-factory.h:
14285 Make element creation more extendible
14286 Add get_element vmethod to the default MediaFactory so that subclasses can just
14287 override that method and still use the default logic for making a MediaBin from
14290 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14293 * gst/rtsp-server/Makefile.am:
14294 * gst/rtsp-server/rtsp-client.c:
14295 * gst/rtsp-server/rtsp-client.h:
14296 * gst/rtsp-server/rtsp-media-factory.c:
14297 * gst/rtsp-server/rtsp-media-factory.h:
14298 * gst/rtsp-server/rtsp-media-mapping.c:
14299 * gst/rtsp-server/rtsp-media-mapping.h:
14300 * gst/rtsp-server/rtsp-media.c:
14301 * gst/rtsp-server/rtsp-media.h:
14302 * gst/rtsp-server/rtsp-server.c:
14303 * gst/rtsp-server/rtsp-server.h:
14304 * gst/rtsp-server/rtsp-session.c:
14305 * gst/rtsp-server/rtsp-session.h:
14306 Make the server handle arbitrary pipelines
14307 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14308 The GstMediaBin object has a handle to a bin with elements and to a list of
14309 GstMediaStream objects that this bin produces.
14310 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14311 with methods to register and remove those mappings.
14312 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14313 used by the server instance.
14314 Modify the example application so that it shows how to create custom pipelines
14315 attached to a specific mount point.
14316 Various misc cleanps.
14318 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14320 * gst/rtsp-server/rtsp-server.c:
14321 * gst/rtsp-server/rtsp-server.h:
14322 Allow setting a custom media factory for a server
14324 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14326 * gst/rtsp-server/rtsp-client.c:
14327 * gst/rtsp-server/rtsp-client.h:
14328 Allow setting a custom media factory for a client.
14330 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14332 * gst/rtsp-server/Makefile.am:
14333 Add Makefile entry for the media factory
14335 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14337 * gst/rtsp-server/rtsp-media-factory.c:
14338 * gst/rtsp-server/rtsp-media-factory.h:
14339 Add media factory to map urls to media pipeline objects.
14341 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14343 * gst/rtsp-server/rtsp-media.c:
14344 * gst/rtsp-server/rtsp-media.h:
14345 Add comments. Remove unused field
14347 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14349 * gst/rtsp-server/rtsp-session-pool.c:
14350 * gst/rtsp-server/rtsp-session-pool.h:
14351 Allow custom session pools to override the session id allocation algorithms Add some comments.
14353 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14355 * gst/rtsp-server/rtsp-session.h:
14358 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14360 * gst/rtsp-server/rtsp-client.c:
14361 * gst/rtsp-server/rtsp-client.h:
14362 Move the connection code in one place Add some comments
14364 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14366 * gst/rtsp-server/rtsp-server.c:
14367 * gst/rtsp-server/rtsp-server.h:
14368 Make vmethod to create and accept new clients. Add some docs.
14370 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14372 * gst/rtsp-server/rtsp-server.c:
14373 * gst/rtsp-server/rtsp-server.h:
14374 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14376 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14378 * gst/rtsp-server/rtsp-client.c:
14379 * gst/rtsp-server/rtsp-client.h:
14380 Name the parameters more appropriately.
14382 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14384 * gst/rtsp-server/rtsp-session-pool.c:
14385 Do some more cleanup of the session pool.
14387 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14389 * gst/rtsp-server/Makefile.am:
14390 * gst/rtsp-server/rtsp-client.c:
14391 Check if return value of gst_rtsp_session_get_media is not NULL
14393 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14395 * gst/rtsp-server/Makefile.am:
14396 Install rtsp-session and rtsp-session-pool headers
14398 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14403 * bindings/python/Makefile.am:
14404 * bindings/python/arg-types.py:
14405 * bindings/python/codegen/Makefile.am:
14406 * bindings/python/codegen/__init__.py:
14407 * bindings/python/codegen/argtypes.py:
14408 * bindings/python/codegen/code-coverage.py:
14409 * bindings/python/codegen/codegen.py:
14410 * bindings/python/codegen/definitions.py:
14411 * bindings/python/codegen/defsparser.py:
14412 * bindings/python/codegen/docextract.py:
14413 * bindings/python/codegen/docgen.py:
14414 * bindings/python/codegen/fileprefix.override:
14415 * bindings/python/codegen/fileprefixmodule.c:
14416 * bindings/python/codegen/h2def.py:
14417 * bindings/python/codegen/mergedefs.py:
14418 * bindings/python/codegen/mkskel.py:
14419 * bindings/python/codegen/override.py:
14420 * bindings/python/codegen/reversewrapper.py:
14421 * bindings/python/codegen/scmexpr.py:
14422 * bindings/python/rtspserver-types.defs:
14423 * bindings/python/rtspserver.defs:
14424 * bindings/python/rtspserver.override:
14425 * bindings/python/rtspservermodule.c:
14427 Add python bindings.
14429 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14431 * bindings/Makefile.am:
14433 Don't go into python dir when requirements for python bindings are missing
14435 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14437 * bindings/Makefile.am:
14438 * bindings/vala/Makefile.am:
14440 Install Vala bindings if vala is available
14442 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14444 * bindings/vala/gst-rtsp-server-0.10.deps:
14445 * bindings/vala/gst-rtsp-server-0.10.vapi:
14446 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14447 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14448 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14449 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14450 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14451 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14452 Regenerated Vala bindings
14454 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14456 * bindings/vala/gst-rtsp-server.vapi:
14457 * bindings/vala/packages/gst-rtsp-server.metadata:
14458 Fixed typo in included headers for vala bindings
14460 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14464 * pkgconfig/Makefile.am:
14465 * pkgconfig/gst-rtsp-server.pc.in:
14466 Added pkgconfig file
14468 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14470 * bindings/vala/gst-rtsp-server.vapi:
14471 * bindings/vala/packages/gst-rtsp-server.excludes:
14472 * bindings/vala/packages/gst-rtsp-server.gi:
14473 * bindings/vala/packages/gst-rtsp-server.metadata:
14474 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14476 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14478 * bindings/vala/gst-rtsp-server.vapi:
14479 * bindings/vala/packages/gst-rtsp-server.deps:
14480 * bindings/vala/packages/gst-rtsp-server.files:
14481 * bindings/vala/packages/gst-rtsp-server.gi:
14482 * bindings/vala/packages/gst-rtsp-server.metadata:
14483 * bindings/vala/packages/gst-rtsp-server.namespace:
14484 Added Vala bindings
14486 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14488 * gst/rtsp-server/rtsp-session.c:
14489 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14491 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14493 * examples/Makefile.am:
14494 * gst/rtsp-server/Makefile.am:
14495 Put GStreamer version in library name
14497 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14499 * examples/Makefile.am:
14500 * gst/rtsp-server/Makefile.am:
14501 Fix some issues to pass distcheck
14503 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14505 * gst/rtsp-server/rtsp-server.c:
14506 Added port property to GstRTSPServer class.
14508 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14513 * examples/Makefile.am:
14516 * gst/rtsp-server/Makefile.am:
14517 * gst/rtsp-server/rtsp-client.c:
14518 * gst/rtsp-server/rtsp-client.h:
14519 * gst/rtsp-server/rtsp-media.c:
14520 * gst/rtsp-server/rtsp-media.h:
14521 * gst/rtsp-server/rtsp-server.c:
14522 * gst/rtsp-server/rtsp-server.h:
14523 * gst/rtsp-server/rtsp-session-pool.c:
14524 * gst/rtsp-server/rtsp-session-pool.h:
14525 * gst/rtsp-server/rtsp-session.c:
14526 * gst/rtsp-server/rtsp-session.h:
14528 Split in library and example program
14530 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14532 * src/rtsp-client.h:
14533 Removed obsolete variable
14535 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14537 * src/rtsp-client.c:
14538 * src/rtsp-client.h:
14539 Removed pipeline variable GstRTSPClient, because it's only used in one function
14541 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14543 * src/rtsp-media.c:
14544 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14546 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14548 * src/rtsp-session.c:
14549 Initialize some more vars.
14551 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14553 * src/rtsp-session.c:
14554 Initialize variable to avoid compiler warning.
14556 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14559 Add a reasonable generic .gitignore