1 2022-09-21 19:19:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4 meson: Use implicit builtin dirs in pkgconfig generation
5 Starting with Meson 0.62, meson automatically populates the variables
6 list in the pkgconfig file if you reference builtin directories in the
7 pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
8 We need this, because ${prefix}/libexec is a hard-coded value which is
9 incorrect on, for example, Debian.
10 Bump requirement to 0.62, and remove version compares that retained
11 support for older Meson versions.
12 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
13 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
15 2021-03-24 14:20:18 -0500 Zebediah Figura <z.figura12@gmail.com>
18 meson: Build with -Wl,-z,nodelete to prevent unloading of dynamic libraries and plugins
19 GLib made the unfortunate decision to prevent libgobject from ever being
20 unloaded, which means that now any library which registers a static type
21 can't ever be unloaded either (and any library that depends on those,
23 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
25 2022-09-05 13:28:18 +1200 Chris Wiggins <chris@chriswiggins.co.nz>
27 * gst/rtsp-server/rtsp-context.c:
28 * gst/rtsp-server/rtsp-context.h:
29 rtsp-server: context: Add method to set the RTSPToken on some RTSPContext
31 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2979>
33 2022-08-24 19:50:19 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
35 * gst/rtsp-server/rtsp-server-internal.h:
36 * gst/rtsp-server/rtsp-stream-transport.c:
37 * gst/rtsp-server/rtsp-stream.c:
38 gst-rtsp-server: Fix pushing backlog to client
39 Check back pressure of a stream transport before popping buffer from its backlog.
40 If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
42 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
44 2022-09-02 16:31:54 +0300 Sebastian Dröge <sebastian@centricular.com>
46 * gst/rtsp-server/rtsp-stream.c:
47 rtsp-server: stream: Don't loop forever if binding to the multicast address fails
48 The address/port is pre-defined by the caller of the function, so
49 retrying is only going to loop forever.
50 Ideally the multicast address should be checked after allocating but
51 this doesn't happen currently, so it's better to error out cleanly then
52 to loop forever trying the same address.
53 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
55 2022-09-01 15:11:31 -0400 Thibault Saunier <tsaunier@igalia.com>
57 * gst/rtsp-sink/meson.build:
59 meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
60 Removing some copy pasted code
61 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
63 2022-09-01 11:51:48 -0400 Thibault Saunier <tsaunier@igalia.com>
66 * gst/rtsp-server/meson.build:
68 meson: Namespace the plugins_doc_dep/libraries variables
69 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
71 2022-08-31 18:44:14 -0400 Thibault Saunier <tsaunier@igalia.com>
74 meson: Rename plugins list and make them "dependency" objects
75 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
77 2022-05-25 18:40:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
79 * gst/rtsp-sink/gstrtspclientsink.c:
80 rtsp+rtmp: Forward warning added to tls-validation-flags to our users
81 With the 2.72 release, glib-networking developers have decided that
82 TLS certificate validation cannot be implemented correctly by them, so
83 they've deprecated it.
84 In a nutshell: a cert can have several validation errors, but there
85 are no guarantees that the TLS backend will return all those errors,
86 and things are made even more complicated by the fact that the list of
87 errors might refer to certs that are added for backwards-compat and
88 won't actually be used by the TLS library.
89 Our best option is to ignore the deprecation and pass the warning onto
90 users so they can make an appropriate security decision regarding
92 We can't deprecate the tls-validation-flags property because it is
93 very useful when connecting to RTSP cameras that will never get
94 updates to fix certificate errors.
95 Relevant upstream merge requests / issues:
96 https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214
97 https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179
98 https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
99 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
101 2022-07-12 16:58:00 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
103 * gst/rtsp-server/rtsp-client.c:
104 rtsp-client: Fix url for generating key in media factory
105 The mount point at / can be accessed by both the URL forms rtsp://<IP>:<PORT> and rtsp://<IP>:<PORT>/.
106 To make media factory generating the same key for both the URL forms, the url sent to gst_rtsp_media_factory_construct() needs to be normalized first.
107 This commit creates a new GstRTSPUrl as the normalized url to send to gst_rtsp_media_factory_construct().
108 Fixes:https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1297
109 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2681>
111 2022-06-29 10:55:13 +0100 Tim-Philipp Müller <tim@centricular.com>
114 coding style: allow declarations after statement
115 See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
116 and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
117 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2683>
119 2022-06-14 16:18:35 +0100 Tim-Philipp Müller <tim@centricular.com>
122 * docs/plugins/gst_plugins_cache.json:
123 * docs/plugins/index.md:
124 * docs/plugins/sitemap.txt:
125 docs: make sure rtspclientsink plugin docs index page is called index.html
126 .. instead of plugin-index.html.
127 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2592>
129 2022-04-06 12:56:30 +0100 Tim-Philipp Müller <tim@centricular.com>
132 Bump GLib requirement to >= 2.62
133 Can't require 2.64 yet because of
134 https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323
135 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
137 2022-05-16 18:06:16 +0200 Patricia Muscalu <patricia@axis.com>
139 * gst/rtsp-server/rtsp-media.c:
140 rtsp-media: Correct logic on GstRTSPStreamBlocking message reception
141 We must take into account the receiving streams as well when calculating
142 the expected number of the received GstRTSPStreamBlocking messages.
143 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2429>
145 2022-04-27 01:13:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
147 * tests/check/gst/onvif.c:
148 tests/onvif: improve robustness
149 The previous iteration of the code was inferring the type of the
150 frame by looking at the overall size of the gst-payloaded packet.
151 It is more robust to actually parse the payload and look at the
152 actual data buffers it contains.
153 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
155 2022-04-27 01:10:46 +0200 Mathieu Duponchelle <mathieu@centricular.com>
157 * tests/check/gst/onvif.c:
158 tests/onvif: don't push buffers outside segment
159 segment->stop is exclusive, so in reverse playback mode we do not
160 need to output a buffer at that position as it will simply get
162 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
164 2022-02-15 13:39:43 +0000 Pierre Bourré <pierre.moltess@gmail.com>
166 * gst/rtsp-sink/gstrtspclientsink.c:
167 rtspclientsink: fix possible shutdown deadlock collect_streams()
168 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1696>
170 2022-04-07 19:14:27 +0300 Sebastian Dröge <sebastian@centricular.com>
172 * gst/rtsp-server/rtsp-sdp.c:
173 rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME
174 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
176 2022-04-13 14:34:57 +0200 Marc Leeman <m.leeman@televic.com>
178 * gst/rtsp-server/rtsp-stream.c:
179 gst-rtsp-server: minor spelling fixes
180 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>
182 2022-03-25 15:00:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
184 * examples/meson.build:
186 Remove glib and gobject dependencies everywhere
187 They are part of gst_dep already and we have to make sure to always have
188 gst_dep. The order in dependencies matters, because it is also the order
189 in which Meson will set -I args. We want gstreamer's config.h to take
190 precedence over glib's private config.h when it's a subproject.
191 While at it, remove useless fallback args for gmodule/gio dependencies,
192 only gstreamer core needs it.
193 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
195 2022-03-28 21:03:16 +1100 Matthew Waters <matthew@centricular.com>
197 * gst/rtsp-server/rtsp-stream.c:
198 rtsp-stream: remove unused variable:
200 ../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
201 guint n_messages = 0;
203 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
205 2022-03-18 13:42:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
208 meson: Bump all meson requirements to 0.60
209 Lots of new warnings ever since
210 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1934
211 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
213 2022-02-23 17:39:18 +0100 Vivienne Watermeier <vwatermeier@igalia.com>
215 * gst/rtsp-server/rtsp-token.c:
216 documentation: improve misleading wording
217 The documentation for several gst_*_writable_structure functions stated
218 that they would never return NULL, without making clear that the passed
219 object is required to be writable. This changes the wording in those
220 cases to make that requirement more clear.
221 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
223 2022-02-10 08:01:02 +0100 Branko Subasic <branko@axis.com>
225 * examples/test-onvif-server.c:
226 * tests/check/gst/onvif.c:
227 rtponviftimestamp: add support for using reference timestamps
228 Make it posible to configure the element to obtain the timestamps from
229 reference timestamp meta data instead of using the ntp-offset property,
230 or estimating its own offset. Currently the only time format supported
231 is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.
232 In addition the custom event GstNtpOffset has been renamed to
233 GstOnvifTimestamp, to reflect that it is not necessarily used to convey
234 the ntp-offset. As a consequence we had to modify a couple of files in
235 the rtsp-server as well.
237 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
239 2022-02-18 16:05:53 +0100 Branko Subasic <branko@axis.com>
241 * tests/check/gst/onvif.c:
242 * tests/check/gst/rtspserver.c:
243 * tests/check/gst/stream.c:
244 gst-rtsp-server: Plug a few memory leaks in tests
245 Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and
246 gst_stream tests by running the tests in valgrind.
247 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
249 2022-03-07 09:14:46 +0100 Branko Subasic <branko@axis.com>
251 * gst/rtsp-server/rtsp-client.c:
252 gst-rtsp-server: fix race in rtsp-client
253 When tunneling over HTTP, if connection on the second channel happens
254 before the control timer is created we may trigger an assert in
255 rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
256 attaching the client thread to the context.
258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
260 2022-02-04 11:15:47 +0000 Tim-Philipp Müller <tim@centricular.com>
262 * docs/gst_plugins_cache.json:
265 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
267 === release 1.20.0 ===
269 2022-02-03 19:53:25 +0000 Tim-Philipp Müller <tim@centricular.com>
274 * docs/gst_plugins_cache.json:
275 * gst-rtsp-server.doap:
279 2022-02-03 19:53:18 +0000 Tim-Philipp Müller <tim@centricular.com>
282 Update ChangeLogs for 1.20.0
284 === release 1.19.90 ===
286 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
291 * docs/gst_plugins_cache.json:
292 * gst-rtsp-server.doap:
296 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
299 Update ChangeLogs for 1.19.90
301 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
303 * examples/test-appsrc2.c:
304 gst-rtsp-server: Fix leak in appsrc2 example
305 In the need-data appsrc callback, a buffer is pulled from the
306 appsink. This buffer is then copied so that metadata is writable.
307 The copy is pushed to the appsrc but it doesn't take ownership
308 of the buffer so we need to manually unref it. The original buffer
309 is finally unreffed when the sample is freed.
310 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
312 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
316 meson: Add explicit check: kwarg to all run_command() calls
317 This is required since Meson 0.61.0, and causes a warning to be
319 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
320 https://github.com/mesonbuild/meson/issues/9300
321 This exposed a bunch of places where we had broken run_command()
322 calls, unnecessary run_command() calls, and places where check: true
324 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
326 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
328 * gst/rtsp-server/meson.build:
329 rtsp-server: add gst_dep to gst_rtsp_server_deps
330 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
331 will avoid the following build failure, because the correct girdir
332 location will be retrieved from gstreamer-1.0.pc:
333 /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
334 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
335 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
336 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
337 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
339 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
340 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
342 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
344 * gst/rtsp-server/rtsp-stream.c:
345 rtsp-stream: fix get_rates raciness
346 Prior to this patch, we considered that a stream was blocking
347 whenever a pad probe was triggered for either the RTP pad or
349 This led to situations where we subsequently unblocked and expected
350 to find a segment on the RTP pad, which was racy.
351 Instead, we now only consider that the stream is blocking when
352 the pad probe for the RTP pad has triggered with a blockable object
353 (buffer, buffer list, gap event).
354 The RTCP pad is simply blocked without affecting the state of the
357 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
359 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
361 * docs/gst_plugins_cache.json:
365 === release 1.19.3 ===
367 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
372 * docs/gst_plugins_cache.json:
373 * gst-rtsp-server.doap:
377 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
380 Update ChangeLogs for 1.19.3
382 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
385 meson: require matching GStreamer dep versions for unstable development releases
386 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
387 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
389 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
391 * tests/check/meson.build:
392 meson: update for meson.build_root() and .build_source() deprecation
393 -> use meson.project_build_root() or .global_build_root() instead.
394 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
396 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
399 * tests/check/meson.build:
400 meson: update for dep.get_pkgconfig_variable() deprecation
401 ... in favour of dep.get_variable('foo', ..) which in some
402 cases allows for further cleanups in future since we can
403 extract variables from pkg-config dependencies as well as
404 internal dependencies using this mechanism.
405 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
407 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
409 * gst/rtsp-server/meson.build:
410 * gst/rtsp-sink/meson.build:
411 rtsp-server: define G_LOG_DOMAIN
413 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
415 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
418 meson: bump meson requirement to >= 0.59
419 For monorepo build and ugly/bad, for advanced feature
420 option API like get_option('xyz').required(..) which
421 we use in combination with the 'gpl' option.
422 For rest of modules for consistency (people will likely
423 use newer features based on the top-level requirement).
424 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
426 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
429 meson: Streamline the way we detect when to build documentation
430 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
432 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
435 * gst/rtsp-server/meson.build:
437 meson: List libraries and their corresponding gir definition
438 Introduces a `libraries` variable that contains all libraries in a
439 list with the following format:
443 'lib': library_object
444 'gir': [ {full gir definition in a dict } ]
449 It therefore refactors the way we build the gir so that we can reuse the
450 same information to build them against 'gstreamer-full' in gst-build
451 when linking statically
452 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
454 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
456 * gst/rtsp-server/meson.build:
457 meson: Mark files as files()
458 Making it more robust and future proof
459 And fix issues that it creates
460 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
462 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
464 * gst/rtsp-server/rtsp-media.c:
465 rtsp-media: Unprepare suspended medias too
466 Previously suspended medias immediately reached the UNPREPARED state
467 without going through the media's unprepare() vfunc. This didn't allow
468 the media subclass to do any additional cleanup, and for example the
469 shutdown-eos property of GstRTSPMedia was ignored.
470 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
472 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
474 * gst/rtsp-server/rtsp-media.c:
475 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
476 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
478 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
480 * gst/rtsp-server/rtsp-client.c:
481 * gst/rtsp-server/rtsp-session.c:
482 * gst/rtsp-server/rtsp-session.h:
483 rtsp-client: make sure sessmedia will not get freed while used
484 handle_*_request() functions were all retrieving the session media from
485 the session by calling gst_rtsp_session_get_media () which is a transfer-none
486 call. If a session timeout happens at that time, the session media may get freed
487 making the pointer invalid..
489 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
491 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
493 * gst/rtsp-server/rtsp-media.c:
494 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
495 Previously the status was only changed for other medias.
496 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
498 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
500 * gst/rtsp-server/rtsp-session.c:
501 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
502 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
503 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
505 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
508 doc: update IRC links to OFTC
509 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
511 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
513 * docs/gst_plugins_cache.json:
516 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
518 === release 1.19.2 ===
520 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
525 * docs/gst_plugins_cache.json:
526 * gst-rtsp-server.doap:
530 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
532 * gst/rtsp-server/rtsp-media.c:
533 * gst/rtsp-server/rtsp-stream.c:
534 * gst/rtsp-server/rtsp-stream.h:
535 * gst/rtsp-sink/gstrtspclientsink.c:
536 Protection against early RTCP packets.
537 When receiving RTCP packets early the funnel is not ready yet and
538 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
539 This causes the thread that handle RTCP packets to go to pause mode.
540 Since this thread is in pause mode there will be no further callbacks to
541 handle keep-alive for incoming RTCP packets. This will make the session
542 time out if the client is not using another keep-alive mechanism.
543 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
544 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
546 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
550 Update COPYING.LIB, COPYING files
551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
553 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
555 * docs/gst_plugins_cache.json:
559 === release 1.19.1 ===
561 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
566 * docs/gst_plugins_cache.json:
567 * gst-rtsp-server.doap:
571 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
573 * gst/rtsp-server/rtsp-stream.c:
574 rtsp-stream: use new gst_buffer_new_memdup()
575 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
577 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
579 * gst/rtsp-server/rtsp-media-factory-uri.c:
580 rtsp-media: fix leak when adding converter
581 Free the previous caps before reusing the variable for the converter caps.
582 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
584 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
586 * gst/rtsp-server/rtsp-client.c:
587 rtsp-client: fix leak adding headers
588 gst_rtsp_message_add_header() makes a copy of the header, instead
590 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
592 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
594 * gst/rtsp-server/rtsp-stream.c:
595 Use gst_element_request_pad_simple...
596 Instead of the deprecated gst_element_get_request_pad.
597 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
599 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
601 * gst/rtsp-server/rtsp-media.c:
602 rtsp-media: Ensure the bus watch is removed during unprepare
603 It's possible for the destruction of the source to be delayed.
604 Instead of relying on the dispose() to remove the bus watch, do
606 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
608 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
611 docs: minor spelling correction in README
612 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
614 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
616 * examples/test-replay-server.c:
617 test-replay-server: minor spelling corrections
618 Bumped on these while investigating the example code.
619 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
621 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
623 * tests/check/gst/stream.c:
624 tests: Don't fail tests if IPv6 not available.
625 On computers with IPv6 disabled it shouldn't result in a test failure.
626 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
628 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
630 * gst/rtsp-server/rtsp-media.c:
631 rtsp-media: Add one more case to seek avoidance
632 This is an extension to the previous commit. There can also be cases where the
633 start position is not specified, in those cases we should also avoid doing
634 seeking unless it's forced.
635 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
637 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
639 * gst/rtsp-server/rtsp-media.c:
640 rtsp-media: Improve skipping trickmode seek.
641 We can also skip the seek if the end range is already
643 Avoids initial seek on play start if playing full stream.
644 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
646 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
648 * gst/rtsp-sink/gstrtspclientsink.c:
649 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
650 It's sufficient to run them during the FIRST stage instead of in both.
651 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
653 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
655 * tests/check/gst/rtspclientsink.c:
656 tests: rtspclientsink: fix some leaks
657 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
659 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
661 * gst/rtsp-sink/gstrtspclientsink.c:
662 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
663 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
665 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
667 * tests/check/gst/rtspclientsink.c:
668 rtspclientsink: add unit test for potential shutdown deadlock
669 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
671 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
673 * gst/rtsp-sink/gstrtspclientsink.c:
674 rtspclientsink: fix deadlock on shutdown before preroll
675 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
676 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
678 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
680 * gst/rtsp-server/rtsp-stream.c:
681 rtsp-stream: avoid deadlock in send_func
682 Currently the send_func() runs in a thread of its own which is started
683 the first time we enter handle_new_sample(). It runs in an outer loop
684 until priv->continue_sending is FALSE, which happens when a TEARDOWN
685 request is received. We use a local variable, cont, which is initialized
686 to TRUE, meaning that we will always enter the outer loop, and at the
687 end of the outer loop we assign it the value of priv->continue_sending.
688 Within the outer loop there is an inner loop, where we wait to be
689 signaled when there is more data to send. The inner loop is exited when
690 priv->send_cookie has changed value, which it does when more data is
691 available or when a TEARDOWN has been received.
692 But if we get a TEARDOWN before send_func() is entered we will get stuck
693 in the inner loop because no one will increase priv->session_cookie
695 By not entering the outer loop in send_func() if priv->continue_sending
696 is FALSE we make sure that we do not get stuck in send_func()'s inner
697 loop should we receive a TEARDOWN before the send thread has started.
698 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
699 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
701 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
703 * gst/rtsp-server/rtsp-client.c:
704 rtsp-client: cleanup transports during TEARDOWN
705 When tunneling RTP over RTSP the stream transports are stored in a hash
706 table in the GstRTSPClientPrivate struct. They are used for, among other
707 things, mapping channel id to stream transports when receiving data from
708 the client. The stream tranports are created and added to the hash table
709 in handle_setup_request(), but unfortuately they are not removed in
710 handle_teardown_request(). This means that if the client sends data on
711 the RTSP connection after it has sent the TEARDOWN, which is often the
712 case when audio backchannel is enabled, handle_data() will still be able
713 to map the channel to a session transport and pass the data along to it.
714 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
715 because the stream is no longer joined to a bin.
716 We avoid this by removing the stream transports from the hash table when
717 we handle the TEARDOWN request.
718 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
720 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
722 * docs/gst_plugins_cache.json:
723 * gst/rtsp-sink/gstrtspclientsink.c:
724 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
725 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
727 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
729 * tests/check/gst/client.c:
730 Add test cases for mountpoint of '/'
731 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
733 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
735 * gst/rtsp-server/rtsp-client.c:
736 * gst/rtsp-server/rtsp-mount-points.c:
737 * gst/rtsp-server/rtsp-session-media.c:
738 Make a mount point of "/" work correctly.
739 As far as I can tell, this is neither explicitly allowed nor
740 forbidden by RFC 7826.
741 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
742 use in the wild (presumably with non-GStreamer servers).
743 GStreamer's prior behavior was confusing, in that
744 gst_rtsp_mount_points_add_factory() would appear to accept a mount
745 path of "" or "/", but later connection attempts would fail with a
746 "media not found" error.
747 This commit makes a mount path of "/" work for either form of URL,
748 while an empty mount path ("") is rejected and logs a warning.
749 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
751 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
753 * docs/gst_plugins_cache.json:
754 * gst/rtsp-sink/gstrtspclientsink.c:
755 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
756 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
758 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
760 * gst/rtsp-server/rtsp-media.c:
761 rtsp-media: Only count senders when counting blocked streams
762 Only sender streams sends the GstRTSPStreamBlocking message, so only
763 these should be counted before setting media status to prepared.
764 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
766 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
768 * gst/rtsp-sink/gstrtspclientsink.c:
769 rtspclientsink add proper support for uri queries
770 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
772 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
774 * gst/rtsp-server/rtsp-client.c:
775 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
776 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
777 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
779 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
781 * gst/rtsp-server/rtsp-stream.c:
782 rtsp-stream: collect a clock_rate when blocking
783 This lets us provide a clock_rate in a fashion similar to the
784 other code paths in get_rtpinfo()
785 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
787 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
789 * gst/rtsp-server/rtsp-media.c:
790 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
791 Otherwise this will cause memory corruption as the property expects a 64
793 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
795 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
797 * gst/rtsp-server/rtsp-media.c:
798 * gst/rtsp-server/rtsp-stream.c:
799 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
800 To prevent cases with prerolling when the inactive stream prerolls first
801 and the server proceeds without waiting for the active stream, we will
802 ignore GstRTSPStreamBlocking messages from incomplete streams. When
803 there are no complete streams (during DESCRIBE), we will listen to all
805 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
807 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
809 * tests/check/gst/media.c:
810 * tests/check/meson.build:
811 * tests/files/test.avi:
812 media test: Add test for seeking one active stream with a demuxer
813 Add another seek_one_active_stream test but with a demuxer. The demuxer
814 will flush both streams in opposed to the existing test which only
815 flushes the active stream. This will help exposing problems with the
816 prerolling process after a flushing seek.
817 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
819 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
821 * gst/rtsp-server/meson.build:
823 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
824 * pkgconfig/gstreamer-rtsp-server.pc.in:
825 * pkgconfig/meson.build:
826 Meson: Use pkg-config generator
827 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
829 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
832 meson: update glib minimum version to 2.56
833 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
835 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
837 * examples/test-launch.c:
838 * gst/rtsp-server/rtsp-media-factory.c:
839 * gst/rtsp-server/rtsp-media-factory.h:
840 * gst/rtsp-server/rtsp-media.c:
841 * gst/rtsp-server/rtsp-server-internal.h:
842 * gst/rtsp-server/rtsp-stream.c:
843 * tests/check/gst/client.c:
844 rtsp-media-factory: expose API to disable RTCP
845 This is supported by the RFC, and can be useful on systems where
846 allocating two consecutive ports is problematic, and RTCP is not
848 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
850 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
852 * hooks/pre-commit.hook:
854 git: use our standard pre commit hook
855 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
857 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
859 * gst/rtsp-server/rtsp-stream.c:
860 rtsp-stream: make use of blocked_running_time in query_position
861 When blocking, the sink element will not have received a buffer
862 yet and the position query will fail. Instead, we make use of
863 the running time of the buffer we blocked on.
864 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
866 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
868 * gst/rtsp-server/rtsp-stream.c:
869 rtsp-stream: collect rtp info when blocking
870 We don't unblock the stream anymore before replying to the
871 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
872 so the sinks don't have a last-sample after potentially flush
873 seeking. seek_trickmode waits for preroll however, which means
874 the stream will block and wait for a first buffer. Subsequent
875 calls to get_rtpinfo() can thus make use of the information.
876 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
877 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
879 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
881 * examples/meson.build:
882 * examples/test-replay-server.c:
883 * examples/test-replay-server.h:
884 examples: Add an example for loop playback
885 This demo example shows a way of file loop playback of a given source.
886 Note that client seek request is not properly implemented yet.
887 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
889 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
891 * gst/rtsp-server/rtsp-media.c:
892 rtsp-media: Plug memory leak
893 The get-storage signal of rtpbin increases the ref count of the storage.
894 So we have to unref it after usage.
895 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
897 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
899 * gst/rtsp-server/rtsp-media.c:
900 rtsp-media: Get rates only on sender streams
901 When play a media with both sender and receiver stream, like ONVIF
902 back channel audio in, gst_rtsp_media_get_rates call
903 gst_rtsp_stream_get_rates for each stream to set the rates. But
904 gst_rtsp_stream_get_rates return false for the receiver steam, which
905 lead a g_assert crash.
906 Instead to get rates on all streams, now just get rates on sender
908 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
910 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
912 * gst/rtsp-server/rtsp-media.c:
913 * gst/rtsp-server/rtsp-server-internal.h:
914 * gst/rtsp-server/rtsp-stream.c:
915 rtsp-media: set a 0 storage size for TCP receivers
916 ulpfec correction is obviously useless when receiving a stream
917 over TCP, and in TCP modes the rtp storage receives non
918 timestamped buffers, causing it to queue buffers indefinitely,
919 until the queue grows so large that sanity checks kick in and
920 warnings start to get emitted.
921 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
923 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
925 * gst/rtsp-server/rtsp-stream.c:
926 rtsp-stream: preroll on gap events
927 This allows negotiating a SDP with all streams present, but only
928 start sending packets at some later point in time
929 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
931 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
933 * gst/rtsp-server/rtsp-media.c:
934 rtsp-media: do not unblock on unsuspend
935 rtsp_media_unsuspend() is called from handle_play_request()
936 before sending the play response. Unblocking the streams here
937 was causing data to be sent out before the client was ready
938 to handle it, with obvious side effects such as initial packets
939 getting discarded, causing decoding errors.
940 Instead we can simply let the media streams be unblocked when
941 the state of the media is set to PLAYING, which occurs after
942 sending the play response.
943 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
945 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
948 ci: include template from gst-ci master branch again
950 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
952 * docs/gst_plugins_cache.json:
956 === release 1.18.0 ===
958 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
964 * docs/gst_plugins_cache.json:
965 * gst-rtsp-server.doap:
969 === release 1.17.90 ===
971 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
976 * docs/gst_plugins_cache.json:
977 * gst-rtsp-server.doap:
981 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
983 * gst/rtsp-server/rtsp-thread-pool.c:
984 rtsp-thread-pool.c: fix clang 10 warning
985 clang 10 is complaining about incompatible types due to the
988 ../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]
990 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
992 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
994 * gst/rtsp-server/rtsp-thread-pool.c:
995 rtsp-thread-pool.c: fix clang 10 warning
996 clang 10 is complaining about incompatible types due to the
999 ../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]
1001 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1003 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
1005 * gst/rtsp-server/rtsp-sdp.c:
1006 rtsp-sdp: Fix resource leak in mikey messsage
1007 Fixed a resource leak for mikey message while adding crypto session
1009 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
1011 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
1014 * scripts/extract-release-date-from-doap-file.py:
1015 meson: set release date from .doap file for releases
1016 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
1018 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1020 * gst/rtsp-server/rtsp-stream.c:
1021 rtsp-stream: explicitly set caps on udpsrc elements
1022 This causes them to send caps events before data flow, which is
1023 usually a pretty correct thing to do!
1024 Not doing so manifested in a bug where ssrcdemux wouldn't forward
1025 the caps it had received with an extra ssrc field, as it hadn't
1026 received any caps event.
1028 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
1030 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1032 * docs/gst_plugins_cache.json:
1036 === release 1.17.2 ===
1038 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1043 * docs/gst_plugins_cache.json:
1044 * gst-rtsp-server.doap:
1048 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
1050 * docs/gst_plugins_cache.json:
1051 doc: Stop documenting properties from parents
1053 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1055 * docs/gst_plugins_cache.json:
1056 docs: Fix version in the plugins cache
1057 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1059 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1061 * gst/rtsp-sink/gstrtspclientsink.c:
1062 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
1063 It's deprecated, unneeded and doesn't do anything anymore.
1064 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1066 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
1071 === release 1.17.1 ===
1073 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
1078 * docs/gst_plugins_cache.json:
1079 * gst-rtsp-server.doap:
1083 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1085 * gst/rtsp-server/rtsp-media.c:
1086 rtsp-media: Add/configure transports when completing the pipeline
1087 Otherwise the transports are not set up yet during the PLAY request
1088 handling when unsuspending (and thus unblocking) the media.
1089 In case of live pipelines this then causes the first few packets to go
1090 to the sinks before they know what to do with them, and they simply
1091 discard them which is rather suboptimal in case of keyframes.
1092 For non-live pipelines this is not a problem because the sink will still
1093 be PAUSED and as such not send out the data yet but wait until it goes
1094 to PLAYING, which is late enough.
1095 Adding the transports multiple times is not a problem: if the transport
1096 is already added it won't be added another time and TRUE will be
1098 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
1100 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
1101 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1103 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1105 * gst/rtsp-server/rtsp-media.c:
1106 rtsp-media: Fix misleading comment
1107 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1109 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
1111 * gst/rtsp-server/rtsp-media.c:
1112 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
1113 The pad probes are not needed anymore at this point and later when
1114 reaching buffering 100% only the state is changed, no unblocking
1116 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1118 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
1120 * gst/rtsp-server/rtsp-media.c:
1121 rtsp-media: Remove duplicated media_unblock() function
1122 It does literally the same as media_streams_set_blocked(FALSE).
1123 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1125 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
1127 * examples/test-onvif-server.c:
1128 test-onvif-server: cast ntp-offset property value to 64 bit
1129 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
1131 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
1133 * docs/gst_plugins_cache.json:
1134 docs: Update plugins cache
1136 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1138 * examples/test-onvif-server.c:
1139 * examples/test-onvif-server.h:
1140 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1141 onvif-media-factory: define autoptr cleanup function
1142 And have the factory in the onvif-server example inherit from
1143 GstRTSPOnvifMediaFactory.
1144 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
1146 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
1148 * docs/gst_plugins_cache.json:
1149 docs: Update plugins cache
1151 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
1153 * tests/check/gst/rtspserver.c:
1154 tests: enforce I420 format
1155 Test was not enforcing a video format on videotestsrc. I420 was picked as it
1156 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
1157 true (gst-plugins-base!689).
1158 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
1160 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1162 * gst/rtsp-sink/gstrtspclientsink.c:
1163 plugins: uddate gst_type_mark_as_plugin_api() calls
1165 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
1168 doc: Require hotdoc >= 0.11.0
1170 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1172 * docs/gst_plugins_cache.json:
1173 docs: Update gst_plugins_cache.json
1175 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1177 * gst/rtsp-sink/gstrtspclientsink.c:
1178 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
1180 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1182 * gst/rtsp-server/meson.build:
1183 meson: gir: remove bogus sources_top_dir kwarg
1184 Doesn't actually exist. Was fixed differently in Meson
1185 so that the user doesn't have to specify it.
1186 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
1188 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1190 * tests/check/meson.build:
1191 tests: put registry into tests/check not the gst/ subdir
1192 Underscorify the test name before setting GST_REGISTRY,
1193 so the registry actually ends up in the current build dir
1194 and not some subdir.
1195 For consistency with the other modules, but should also
1196 avoid problems on windows.
1197 Also fix indentation of environment block.
1198 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1200 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
1202 * tests/check/meson.build:
1203 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
1204 If core is built as a subproject (e.g. as in gst-build), make sure to use
1205 the gst-plugin-scanner from the built subproject. Without this, gstreamer
1206 might accidentally use the gst-plugin-scanner from the install prefix if
1207 that exists, which in turn might drag in gst library versions we didn't
1208 mean to drag in. Those gst library versions might then be older than
1209 what our current build needs, and might cause our newly-built plugins
1210 to get blacklisted in the test registry because they rely on a symbol
1211 that the wrongly-pulled in gst lib doesn't have.
1212 This should fix running of unit tests in gst-build when invoking
1213 meson test or ninja test from outside the devenv for the case where
1214 there is an older or different-version gst-plugin-scanner installed
1215 in the install prefix.
1216 In case no gst-plugin-scanner is installed in the install prefix, this
1217 will fix "GStreamer-WARNING: External plugin loader failed. This most
1218 likely means that the plugin loader helper binary was not found or
1219 could not be run. You might need to set the GST_PLUGIN_SCANNER
1220 environment variable if your setup is unusual." warnings when running
1222 In the case where we find GStreamer core via pkg-config we use
1223 a newly-added pkg-config var "pluginscannerdir" to get the right
1224 directory. This has the benefit of working transparently for both
1225 installed and uninstalled pkg-config files/setups.
1226 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1228 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
1230 * tests/check/meson.build:
1231 tests: gst-plugins-base and -bad plugins are required for the unit tests
1232 Make hard requirement until we have more fine-grained control
1233 in the unit tests. Of course the presence of the .pc file doesn't
1234 imply that the plugins we need are actually there, but it's at
1235 least a step in the right direction.
1236 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1238 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1240 * tests/check/meson.build:
1241 tests: pick up rtsp-server plugins from build directory only
1242 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1244 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
1246 * gst/rtsp-server/rtsp-media.c:
1247 rtsp-media: wait for all GstRTSPStreamBlocking messages
1248 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
1249 each active stream when checking if all streams are blocked.
1250 Without this change there will be a race condition when using two or
1251 more streams and rtsp-media receives a GstRTSPStreamBlocking message
1252 from one of the streams. This is because rtsp-media then checks if all
1253 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
1254 stream. This function call returns TRUE if the stream has sent a
1255 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
1256 receive this message. This would then result in that rtsp-media
1257 erroneously thinks it is blocking all streams which could result in
1258 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
1259 preroll, this could result in that rtsp-media thinks that the pipeline
1260 is prerolled even though that might not be the case.
1261 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
1263 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
1265 * gst/rtsp-server/rtsp-media.c:
1266 rtsp-media: update expected_async_done during suspend
1267 Set expected_async_done to FALSE in default_suspend() if a state change
1268 occurs and the return value from set_target_state() is something other
1269 than GST_STATE_CHANGE_ASYNC.
1270 Without this change there is a risk that expected_async_done will be
1271 TRUE even though no asynchronous state change is taking place. This
1272 could happen if the pipeline is set to PAUSED using
1273 media_set_pipeline_state_locked(), an asynchronous state change starts
1274 and then the media is suspended (which could result in a state change,
1275 aborting the asynchronous state change). If the media is suspended
1276 before the asynchronous state change ends then expected_async_done will
1277 be TRUE but no asynchronous state change is taking place.
1278 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
1280 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
1282 * gst/rtsp-server/rtsp-client.c:
1283 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1284 There was a race condition where client was being finalized and
1285 concurrently in some other thread the rtsp ctrl timout was relying on
1286 client data that was being freed.
1287 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1288 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1290 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1292 * gst/rtsp-server/rtsp-media-factory.c:
1293 * gst/rtsp-server/rtsp-media-factory.h:
1294 * gst/rtsp-server/rtsp-media.c:
1295 * gst/rtsp-server/rtsp-media.h:
1296 media-factory: complete DSCP QoS setting support
1297 add dscp_qos setting support at factory and media level to setup IP DSCP
1298 field of bounded UDP sinks.
1299 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1300 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1302 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1304 * gst/rtsp-server/rtsp-client.c:
1305 rtsp-client: Fix some race conditions around timeout source removal
1306 We always need to take the lock while accessing it as otherwise another
1307 thread might've removed it in the meantime. Also when destroying and
1308 creating a new one, ensure that the mutex is not shortly unlocked in
1309 between as during that time another one might potentially be created
1311 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1313 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1315 * gst/rtsp-server/rtsp-media.c:
1316 * gst/rtsp-server/rtsp-stream.c:
1317 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1318 And the same for gst_rtsp_stream_get_rates().
1319 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1321 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1323 * examples/test-onvif-server.c:
1324 examples: test-onvif-server: fix compiler warnings on raspbian
1325 Fix printf format for 64-bit variables.
1326 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1328 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1330 * gst/rtsp-server/rtsp-stream-transport.c:
1331 * gst/rtsp-server/rtsp-stream-transport.h:
1332 * gst/rtsp-server/rtsp-stream.c:
1333 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1334 The old API is preserved now and new API was added that provides the
1335 additional parameter to the callback.
1336 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1337 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1339 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1341 * gst/rtsp-server/rtsp-client.c:
1342 rtsp-client: Store the timeout source by pointer instead of id
1343 That way we don't have to retrieve it again from the main context when
1344 destroying it but can directly do so.
1345 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1347 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1349 * gst/rtsp-server/rtsp-client.c:
1350 rtsp-client: Clean up watch/watch context and related state consistently
1351 And assert that it was cleaned up properly before the client is
1352 finalized. If something is still around when the client is shut down
1353 then something went very wrong before.
1354 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1356 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1358 * gst/rtsp-server/rtsp-client.c:
1359 * tests/check/gst/rtspserver.c:
1360 rtsp-client: Combine the pre-session and post-session timeout
1361 They previously used the same state but different mechanisms and
1362 functions, which was difficult to follow, error prone and simply
1364 Also adjust the test for the post-session timeout a bit to be less racy
1365 now that the timing has slightly changed.
1366 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1368 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1370 * gst/rtsp-server/rtsp-client.c:
1371 rtsp-client: Don't ever close the client connection directly when a session is torn down
1372 There might be other sessions that are running over the same RTSP
1373 connection and we should not simply close the client directly if one of
1375 By default the connection will be closed once the client closes it or
1376 the OS does. This behaviour can be adjusted with the
1377 post-session-timeout property, which allows to close it automatically
1378 from the server side after all sessions are gone and the given timeout
1380 This reverts the previous commit.
1381 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1383 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1385 * gst/rtsp-server/rtsp-client.c:
1386 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1387 Instead of closing it never at all. Previously there was only code that
1388 closed the client asynchronously if sending the response happened
1389 asynchrously at a later time.
1390 Thanks to Christian M for debugging this issue.
1391 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1392 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1394 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1396 * gst/rtsp-server/rtsp-stream.c:
1397 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1398 Otherwise no sink is found for multicast sreams and the less accurate
1399 fallback is used to determine the current sequence number and timestamp.
1401 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1403 * gst/rtsp-server/rtsp-auth.c:
1404 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1405 When using the basic authentication scheme, we wouldn't validate that
1406 the authorization field of the credentials is not NULL and pass it on
1407 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1408 dereference the NULL pointer and crash.
1409 A specially crafted (read: invalid) RTSP header can cause this to
1411 As a solution, check for the authorization to be not NULL before
1412 continuing processing it and if it is simply fail authentication.
1413 This fixes CVE-2020-6095 and TALOS-2020-1018.
1414 Discovered by Peter Wang of Cisco ASIG.
1416 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1418 * gst/rtsp-server/rtsp-client.c:
1419 rtsp-client: Use watch_context before unref
1420 Move the usage of priv->watch_context to beginning of function
1421 gst_rtsp_client_finalize. Instead of use it after
1422 g_main_context_unref (priv->watch_context).
1424 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1426 * gst/rtsp-server/rtsp-stream.c:
1427 rtsp-stream: fix deadlock on transport removal
1428 We cannot take the RTSPStream lock while holding a transport backlog
1429 lock, as remove_transport may be called externally, which will
1430 take first the RTSPStream lock then the transport backlog lock.
1432 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1434 * gst/rtsp-server/rtsp-server-internal.h:
1435 * gst/rtsp-server/rtsp-stream-transport.c:
1436 * gst/rtsp-server/rtsp-stream.c:
1437 rtsp-stream: clear backlog when removing transport
1438 This ensures we don't end up calling any of transports' callbacks
1439 with a potentially unreffed user_data (in practice, a client that
1440 may have been removed)
1442 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1444 * gst/rtsp-server/rtsp-stream.c:
1445 rtsp-stream: marshal calls to send_tcp_message to a single thread
1446 In order to address the race condition pointed out at
1447 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1448 we get rid of the send thread pool, and instead spawn and manage
1449 a single thread to pull samples from app sinks and add them to
1450 the transport's backlogs.
1451 Additionally, we now also always go through the backlogs in order
1452 to simplify the logic.
1454 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1456 * gst/rtsp-server/rtsp-server-internal.h:
1457 * gst/rtsp-server/rtsp-stream-transport.c:
1458 * gst/rtsp-server/rtsp-stream.c:
1459 rtsp-stream: properly protect TCP backlog access
1461 We cannot hold stream->lock while pushing data, but need
1462 to consistently check the state of the backlog both from
1463 the send_tcp_message function and the on_message_sent function,
1464 which may or may not be called from the same thread.
1465 This commit introduces internal API to allow for potentially
1466 recursive locking of transport streams, addressing a race
1467 condition where the RTSP stream could push items out of order
1468 when popping them from the backlog.
1470 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1472 * gst/rtsp-server/rtsp-media.c:
1473 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1474 It's taken ownership of by the media, and returned with `transfer none`
1475 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1476 first then any bindings will wrongly take ownership of the pipeline once
1477 it arrives in bindings code.
1479 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1481 * examples/test-onvif-client.c:
1482 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1484 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1486 * gst/rtsp-server/rtsp-media.c:
1487 rtsp-media: fix default latency
1489 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1491 * gst/rtsp-server/rtsp-client.c:
1492 rtsp-client: make closing more thread safe
1493 + Take the watch lock prior to using priv->watch
1494 + Flush both the watch and connection before closing / unreffing
1495 gst_rtsp_connection_close() is not threadsafe on its own, this is
1496 a workaround at the client level, where we control both the watch
1499 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1501 * gst/rtsp-server/rtsp-latency-bin.c:
1502 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1505 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1506 `your_type_get_instance_private()` function instead
1509 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1511 * gst/rtsp-server/rtsp-client.c:
1512 * tests/check/gst/rtspserver.c:
1513 rtsp-client: add property post-session-timeout
1514 This is a TCP connection timeout for client connections, in seconds.
1515 If a positive value is set for this property, the client connection
1516 will be kept alive for this amount of seconds after the last session
1517 timeout. For negative values of this property the connection timeout
1518 handling is delegated to the system (just as it was before).
1521 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1523 * gst/rtsp-server/rtsp-stream.c:
1524 rtsp-stream: check for NULL transports prior to ref'ing
1526 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1528 * gst/rtsp-server/rtsp-server-internal.h:
1529 * gst/rtsp-server/rtsp-stream-transport.c:
1530 * gst/rtsp-server/rtsp-stream.c:
1531 rtsp-stream: fix checking of TCP backpressure
1532 The internal index of our appsinks, while it can be used to
1533 determine whether a message is RTP or RTCP, is not necessarily
1534 the same as the interleaved channel. Let the stream-transport
1535 determine the channel to check backpressure for, the same way
1536 it determines the channel according to whether it is sending
1539 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1541 * gst/rtsp-server/rtsp-session.c:
1542 rtsp-session: Butcher the file to please gst-indent in the CI
1543 This should be reverted once the CI has an updated gst-indent.
1545 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1547 * gst/rtsp-server/rtsp-session.c:
1548 * gst/rtsp-server/rtsp-session.h:
1549 * gst/rtsp-sink/gstrtspclientsink.c:
1550 * gst/rtsp-sink/gstrtspclientsink.h:
1551 rtsp-session & client: Remove deprecated GTimeVal
1552 GTimeVal won't work past 2038
1554 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1556 * gst/rtsp-server/rtsp-auth.c:
1557 rtsp-auth: fix default token leak
1559 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1561 * gst/rtsp-sink/gstrtspclientsink.c:
1562 gstrtspclientsink: unref transports when closing bin
1565 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1567 * gst/rtsp-server/rtsp-media.c:
1568 rtsp-media: Force seek when flush flag is set
1569 The commit "rtsp-client: define all seek accuracy flags from
1570 setup_play_mode" changed the behaviour of when doing a seek.
1571 Before that commit, having the flush flag set would result in a seek
1573 Even if no seek was needed. One reason to force seek is to flush old buffers
1574 created in Describe requests.
1575 Thus adding force seek also for flush flag will result in play request
1578 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1580 * gst/rtsp-server/rtsp-client.c:
1581 rtsp-client: Revitalize dead code
1582 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1585 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1587 * gst/rtsp-server/rtsp-sdp.c:
1588 rtsp-sdp: Don't try to use non-initialized values
1589 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1590 returns TRUE. Also avoid the whole clock signalling block if we're not
1591 dealing with senders.
1596 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1598 * gst/rtsp-server/rtsp-stream-transport.c:
1599 * gst/rtsp-server/rtsp-stream.c:
1600 * tests/check/gst/stream.c:
1601 rtsp-stream: Removing invalid transports returns false
1602 When removing transports an assertion was that the transports passed in
1603 for removal are present in the list, however that can't be assumed.
1604 As an example if a transport was removed from a thread running
1605 send_tcp_message, the main thread can try to remove the same transport
1606 again if it gets a handle_pause_request. This will not effect the
1607 transport list but it will effect n_tcp_transports as it will be
1608 decrement and then have the wrong value.
1610 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1612 * tests/check/gst/client.c:
1613 client test: add scale and speed negative tests
1614 Negative tests for scale and speed should be done as well, verify that
1615 the response code is "400 Bad request" when a bad request is done.
1617 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1619 * gst/rtsp-server/rtsp-auth.c:
1620 * gst/rtsp-server/rtsp-client.c:
1621 * gst/rtsp-server/rtsp-media-factory.c:
1622 * gst/rtsp-server/rtsp-media.c:
1623 * gst/rtsp-server/rtsp-server.c:
1624 * gst/rtsp-server/rtsp-session-pool.c:
1625 * gst/rtsp-server/rtsp-stream.c:
1626 * gst/rtsp-sink/gstrtspclientsink.c:
1627 Don't pass default GLib marshallers for signals
1628 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1629 actually internally optimize the signal (if the marshaller is available
1630 in GLib itself) by also setting the valist marshaller. This makes the
1631 signal emission a bit more performant than the regular marshalling,
1632 which still needs to box into `GValue` and call libffi in case of a
1634 Note that for custom marshallers, one would use
1635 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1637 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1639 * gst/rtsp-server/rtsp-mount-points.c:
1640 GstRTSPMountPoints: Remove any existing factory before adding a new one
1641 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1642 previous mount point will be freed" which was true when it was
1643 implemented using a GHashTable. But in 2012 it got rewrote using a
1644 GSequence and since then it could have 2 factories for the same path.
1645 Which one gets used is random, depending on the sorting order of 2
1648 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1650 * gst/rtsp-server/rtsp-client.c:
1651 * gst/rtsp-server/rtsp-server-internal.h:
1652 * gst/rtsp-server/rtsp-stream-transport.c:
1653 * gst/rtsp-server/rtsp-stream-transport.h:
1654 * gst/rtsp-server/rtsp-stream.c:
1655 stream: refactor TCP backpressure handling
1656 The previous implementation stopped sending TCP messages to
1657 all clients when a single one stopped consuming them, which
1658 obviously created problems for shared media.
1659 Instead, we now manage a backlog in stream-transport, and slow
1660 clients are removed once this backlog exceeds a maximum duration,
1661 currently hardcoded.
1664 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1667 meson: build gir even when cross-compiling if introspection was enabled explicitly
1668 This can be made to work in certain circumstances when
1669 cross-compiling, so default to not building g-i stuff
1670 when cross-compiling, but allow it if introspection was
1671 enabled explicitly via -Dintrospection=enabled.
1672 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1674 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1676 * gst/rtsp-server/rtsp-session.c:
1677 rtsp-session: clean up comment extra-timeout
1679 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1681 * gst/rtsp-server/rtsp-client.c:
1682 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1683 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1684 from the RTSP context.
1687 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1689 * gst/rtsp-server/rtsp-client.c:
1690 * gst/rtsp-server/rtsp-media.c:
1691 * gst/rtsp-server/rtsp-media.h:
1692 rtsp-client: Lock shared media
1693 For shared media we got race conditions. Concurrently rtsp clients might
1694 suspend or unsuspend the shared media and thus change the state without
1695 the clients expecting that.
1696 By introducing a lock that can be taken by callers such as rtsp_client
1697 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1698 to handle the media sequentially thus allowing one client to finish its
1699 rtsp call before another client calls on the same media.
1700 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1703 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1705 * gst/rtsp-server/rtsp-session.c:
1706 rtsp-session: add property extra-timeout
1707 Extra time to add to the timeout, in seconds. This only
1708 affects the time until a session is considered timed out
1709 and is not signalled in the RTSP request responses.
1710 Only the value of the timeout property is signalled in the
1713 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1715 * gst/rtsp-server/rtsp-stream.c:
1716 rtsp-stream : fix race condition in send_tcp_message
1717 If one thread is inside the send_tcp_message function and are done
1718 sending rtp or rtcp messages so the n_outstanding variable is zero
1719 however have not exit the loop sending the messages. While sending its
1720 messages, transports have been added or removed to the transport list,
1721 so the cache should be updated. If now an additional thread comes to
1722 the function send_tcp_message and trying to send rtp messages it will
1723 first destroy the rtp cache that is still being iterated trough by the
1727 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1736 * examples/.gitignore:
1737 * examples/Makefile.am:
1739 * gst/rtsp-server/.gitignore:
1740 * gst/rtsp-server/Makefile.am:
1741 * gst/rtsp-sink/Makefile.am:
1742 * pkgconfig/.gitignore:
1743 * pkgconfig/Makefile.am:
1745 * tests/Makefile.am:
1746 * tests/check/Makefile.am:
1747 Remove autotools build
1749 Maybe we can now use the meson pkgconfig module
1750 for .pc files? (Does it support uninstalled now?)
1752 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1754 * tests/check/gst/client.c:
1755 client: fix test mem leak in attach_rate_tweaking_probe
1757 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1759 * tests/check/gst/media.c:
1760 media: remove memleak in test test_media_seek
1762 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1764 * tests/check/gst/rtspserver.c:
1765 rtspserver: Remove memleak in test test_double_play
1767 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1769 * gst/rtsp-server/rtsp-media.c:
1770 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1772 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1774 * gst/rtsp-server/rtsp-media.c:
1775 * tests/check/gst/rtspserver.c:
1776 rtsp-media: Unblock all streams
1777 When unsuspending and going to PLAYING, unblock all streams instead of
1778 only those that are linked (the linked streams are the ones for which
1779 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1780 pushing buffers on unlinked streams.
1781 This change is because playback using single-threaded demuxers like
1782 matroska-demux could be blocked if SETUP was not called for all media.
1783 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1784 gstflvdemux, qtdemux, and matroska-demux) will handle
1785 GST_FLOW_NOT_LINKED automatically.
1788 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1790 * gst/rtsp-server/rtsp-media.c:
1791 * tests/check/gst/rtspserver.c:
1792 rtsp-media: Wait on async when needed.
1793 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1794 In the unit test the pause from adjust_play_mode will cause a preroll
1795 and after that async-done will be produced.
1796 Without this patch there are no one consuming this async-done and when
1797 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1798 wait for async-done. But then it wrongly find the async-done prodused by
1799 adjus_play_mode and continue executing without waiting for the preroll
1802 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1804 * gst/rtsp-server/rtsp-client.c:
1805 rtsp-client: RTP Info when completed_sender
1806 Change condition that should be fulfilled regarding RTPInfo.
1807 Replace !gst_rtsp_media_is_receive_only with
1808 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1809 for a sender pipeline that is complete. Only then a RTPInfo should
1811 gst_rtsp_media_is_receive_only gives different answears depending on
1813 If Describe is called wth URL+options for backchannel SDP will give only
1814 audio and only backchannel a=sendonly
1815 If Describe is called on URL+options that gives both audio and video
1816 direction from server to client, pipelines are created. Thus
1817 receive_only will return false, even though Setup only would setup
1819 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1820 streams are complete.
1822 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1824 * gst/rtsp-server/rtsp-client.c:
1825 * tests/check/gst/client.c:
1826 rtsp-client: RTP Info exists conditionally in PLAY
1827 If RTP Info is missing and it is not a receiver only, eg. audio
1828 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1829 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1830 Since 1.14 there is audio backchannel support. Thus RTP-info is
1831 conditional now. When audio backchannel only mode, there is no RTP-info.
1834 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1836 * examples/test-onvif-client.c:
1837 test-onvif-client: remove unused query
1839 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1841 * gst/rtsp-server/rtsp-client.c:
1842 rtsp-client: RTP Info must exist in PLAY response
1843 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1846 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1848 * examples/test-onvif-client.c:
1849 test-onvif-client: perform accurate seeks
1850 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1851 Also, modify how we compute the position: position queries in
1852 PAUSED mode fail to account for the newly-prerolled frame, leading
1853 to frame skips when performing seeks in that state. Instead,
1854 compute the current position from the last sample.
1856 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1858 * gst/rtsp-server/rtsp-client.c:
1859 * gst/rtsp-server/rtsp-media.c:
1860 * gst/rtsp-server/rtsp-media.h:
1861 * tests/check/gst/rtspserver.c:
1862 Use complete streams for scale and speed.
1863 Without this patch it's always stream0 that is used to get segment event
1864 that is used to set scale and speed. This even if client not doing SETUP
1865 for stream0. At least in suspend mode reset this not working since then
1866 it's just random if send_rtp_sink have got any segment event. There are
1867 no check if send_rtp_sink for stream0 got any data before media is
1868 prerolled after PLAY request.
1870 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1872 * examples/test-onvif-server.c:
1873 * examples/test-onvif-server.h:
1874 examples/onvif-server: fix werror build with clang
1875 ../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]
1876 self->incoming_segment->format, self->incoming_segment->flags,
1877 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1878 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1879 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1881 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1882 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1884 <scratch space>:77:1: note: expanded from here
1887 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1888 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1890 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1891 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1893 <scratch space>:9:1: note: expanded from here
1896 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1897 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1898 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1900 <scratch space>:12:1: note: expanded from here
1904 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1907 meson: Don't generate doc cache when no plugins are enabled
1908 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1910 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1912 * examples/test-onvif-client.c:
1913 test-onvif-client: stdin is not defined in MSVC
1915 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1917 * gst/rtsp-server/rtsp-media.c:
1918 rtsp-media: add missing Since tag
1920 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1922 * examples/test-onvif-client.c:
1923 test-onvif-client: STDIN_FILENO is not portable
1924 If not defined, define it to _fileno(stdin) on Windows, 0
1927 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1929 * examples/test-onvif-server.c:
1930 test-onvif-server: downgrade logging
1932 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1934 * examples/meson.build:
1935 * examples/test-onvif-client.c:
1936 * examples/test-onvif-server.c:
1937 examples: add ONVIF client / server example
1939 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1941 * gst/rtsp-server/rtsp-client.c:
1942 * gst/rtsp-server/rtsp-media.c:
1943 rtsp-client: define all seek accuracy flags from setup_play_mode
1944 We then pass those to adjust_play_mode, which needs to operate
1945 on the "final" seek flags, as previously the code in rtsp-media
1946 was assuming that accuracy seek flags (accurate / key_unit) should
1947 not be set if the flags passed to the seek method were already set.
1949 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1951 * gst/rtsp-server/rtsp-media-factory-uri.c:
1952 * gst/rtsp-server/rtsp-media.c:
1953 rtsp-media: Try to get dynamic payloaders by name from their bin first
1954 First try "pay", then "pay_%s" (where %s == pad name). And only then
1955 fall back to the code that simply takes the first payloader that is
1957 The current code usually works (but is racy) because it will always take
1958 the payloader that was last added (due to g_list_prepend() when adding
1959 elements) in pad-added and that's usually the correct one. But if a new
1960 payloader is added between pad-added and us trying to get it, we would
1961 get the wrong payloader.
1963 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1965 * tests/check/gst/client.c:
1966 client test: expect any port in transport
1967 setup_multicast_client sets a 5000-5010 range for the client
1968 ports, it is incorrect to expect the transport to always use
1972 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1974 * tests/check/gst/onvif.c:
1975 onvif tests: use g_cond_wait() correctly
1976 g_cond_wait() has to be called in a loop until required conditions
1980 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1982 * gst/rtsp-server/rtsp-stream.c:
1983 rtsp-stream: Not wait on receiver streams when pre-rolling
1984 Without this patch there are problem pre-rolling when using audio back
1986 Without this patch a probe will be created for all streams including
1987 the stream for audio backchannel. To pre-roll all this pads have to
1988 receive data. Since the stream for audio backchannel is a receiver this
1990 The solution is to never create any probes for streams that are for
1991 incomming data and instead set them as blocking already from beginning.
1993 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1995 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1996 * gst/rtsp-server/rtsp-onvif-media.c:
1997 onvif-media: fix "void function returning a value" compiler warning
1999 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2001 * gst/rtsp-server/rtsp-media.c:
2002 rtsp-media: make sure streams are blocked when sending seek
2003 The recent ONVIF work exposed a race condition when dealing with
2004 multiple streams: one of the sinks may preroll before other streams
2005 have started flushing. This led to the pipeline posting async-done
2006 prematurely, when some streams were actually still in the middle
2007 of performing a flushing seek. The newly-added code looks up a
2008 sticky segment event on the first stream in order to respond to
2009 the PLAY request with accurate Scale and Speed headers. In the
2010 failure condition, the first stream was flushing, and thus had
2011 no sticky segment event, leading to the PLAY request failing,
2012 and in turn the test.
2014 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
2017 * gst/rtsp-server/rtsp-media-factory-uri.h:
2020 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2022 * gst/rtsp-server/rtsp-client.c:
2023 * gst/rtsp-server/rtsp-client.h:
2024 * gst/rtsp-server/rtsp-media.c:
2025 * gst/rtsp-server/rtsp-media.h:
2026 * gst/rtsp-server/rtsp-onvif-client.c:
2027 * gst/rtsp-server/rtsp-onvif-client.h:
2028 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2029 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2030 * gst/rtsp-server/rtsp-onvif-media.c:
2031 * gst/rtsp-server/rtsp-onvif-server.h:
2032 * gst/rtsp-server/rtsp-stream.c:
2033 * gst/rtsp-server/rtsp-stream.h:
2034 * tests/check/gst/media.c:
2035 * tests/check/gst/onvif.c:
2036 * tests/check/meson.build:
2037 onvif: Implement and test the Streaming Specification
2038 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2040 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2042 * gst/rtsp-server/rtsp-client.c:
2043 * gst/rtsp-server/rtsp-client.h:
2044 rtsp-client: add gst_rtsp_client_get_stream_transport()
2045 This will be used in the onvif tests in order to validate the
2046 data transmitted over TCP: for streaming to continue after a
2047 data message has been provided to client->send_func, the client
2048 is responsible for marking the message as sent on the relevant
2051 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2053 * gst/rtsp-server/rtsp-client.c:
2054 client: Scale implies TRICK_MODE
2056 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2058 * gst/rtsp-server/rtsp-client.c:
2059 client: compare booleans, not pointers to them
2061 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
2063 * gst/rtsp-server/rtsp-media.c:
2064 * gst/rtsp-server/rtsp-stream.c:
2065 * tests/check/gst/media.c:
2066 Reverse playback support
2067 GStreamer plays segment from stop to start when doing reverse playback.
2068 RTSP implies that media should be played from start of Range header to
2069 its stop. Hence we swap start and stop times before passing them to
2071 Also make gst_rtsp_stream_query_stop always return value that can be
2072 used as stop time of Range header.
2074 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
2076 * gst/rtsp-server/rtsp-client.c:
2077 * gst/rtsp-server/rtsp-media.c:
2078 * gst/rtsp-server/rtsp-media.h:
2079 * tests/check/gst/client.c:
2080 rtsp-client: add support for Scale and Speed header
2081 Add support for the RTSP Scale and Speed headers by setting the rate in
2082 the seek to (scale*speed). We then check the resulting segment for rate
2083 and applied rate, and use them as values for the Speed and Scale headers
2085 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2087 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
2089 * gst/rtsp-server/rtsp-client.c:
2090 * gst/rtsp-server/rtsp-client.h:
2091 rtsp-client: allow sub classes to adjust the seek
2092 Adds a new virtual function, adjust_play_mode(), that allows
2093 sub classes to adjust the seek done on the media. The sub class can
2094 modify the values of the the seek flags and the rate.
2095 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2097 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
2099 * gst/rtsp-server/rtsp-media.c:
2100 * gst/rtsp-server/rtsp-media.h:
2101 * gst/rtsp-server/rtsp-stream.c:
2102 * gst/rtsp-server/rtsp-stream.h:
2103 * tests/check/gst/media.c:
2104 rtsp-media: allow specifying rate when seeking
2105 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
2106 caller to specify the rate for the seek. Also added functions in
2107 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
2108 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2110 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
2114 meson: Bump minimal GLib version to 2.44
2115 This means we can use some newer features and get rid of some
2116 boilerplate code using the G_DECLARE_* macros.
2117 As discussed on IRC, 2.44 is old enough by now to start depending on it.
2119 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2121 * docs/libs/.gitignore:
2122 * docs/libs/Makefile.am:
2123 * docs/libs/gst-rtsp-server-docs.sgml:
2124 * docs/libs/gst-rtsp-server-sections.txt:
2125 * docs/libs/gst-rtsp-server.types:
2126 docs: remove obsolete gtk-doc related files
2128 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2130 * gst/rtsp-sink/gstrtspclientsink.c:
2131 doc: remove xml from comments
2133 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
2135 * docs/gst_plugins_cache.json:
2137 docs: Stop building the doc cache by default
2138 And update the cache
2139 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
2141 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
2143 * docs/gst_plugins_cache.json:
2144 docs: Update plugins documentation cache
2146 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
2149 * gst/rtsp-server/rtsp-context.c:
2150 * gst/rtsp-server/rtsp-session-pool.c:
2151 doc: Fix some docstrings
2153 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
2159 * docs/gst_plugins_cache.json:
2162 * docs/plugin-index.md:
2163 * docs/plugin-sitemap.txt:
2166 * docs/version.entities.in:
2167 * gst/rtsp-server/meson.build:
2168 * gst/rtsp-sink/meson.build:
2170 * meson_options.txt:
2171 docs: Port to hotdoc
2173 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2175 * gst/rtsp-server/rtsp-auth.c:
2176 * gst/rtsp-server/rtsp-client.h:
2177 rtsp-server: Fix various Since markers
2179 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
2181 * gst/rtsp-server/rtsp-media.c:
2182 * gst/rtsp-server/rtsp-sdp.c:
2183 * gst/rtsp-server/rtsp-session-media.c:
2184 * gst/rtsp-server/rtsp-stream.c:
2185 rtsp-server: Add various Since: 1.14 markers
2187 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
2189 * gst/rtsp-server/rtsp-media-factory.c:
2190 * gst/rtsp-server/rtsp-media.c:
2191 * gst/rtsp-server/rtsp-stream-transport.c:
2192 * gst/rtsp-server/rtsp-stream.c:
2193 rtsp-server: Add various missing Since: 1.16 markers
2195 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
2197 * gst/rtsp-sink/gstrtspclientsink.c:
2198 rtspclientsink: Set async-handling=false for the internal bins
2199 Without this we can easily run into a race condition with async state changes:
2200 - the pipeline is doing an async state change
2201 - we set the internal bins to PLAYING but that's ignored because an
2202 async state change is currently pending
2203 - the async state change finishes but does not change the state of the
2204 internal bins because of locked_state==TRUE
2205 - the internal bins stay in PAUSED forever
2207 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2209 * gst/rtsp-sink/gstrtspclientsink.c:
2210 rtspclientsink: Use write_messages() API to send buffer lists in one go
2211 And to write messages with multiple memories also via writev().
2213 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
2215 * gst/rtsp-server/rtsp-client.c:
2216 * gst/rtsp-server/rtsp-client.h:
2217 * gst/rtsp-server/rtsp-server-object.h:
2218 * gst/rtsp-server/rtsp-server.c:
2219 rtsp-client: Handle Content-Length limitation
2220 Add functionality to limit the Content-Length.
2221 API addition, Enhancement.
2222 Define an appropriate request size limit and reject requests
2223 exceeding the limit with response status 413 Request Entity Too Large
2226 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2233 === release 1.16.0 ===
2235 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2241 * gst-rtsp-server.doap:
2245 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
2247 * gst/rtsp-sink/gstrtspclientsink.c:
2248 rtspclientsink: Notify the stream transport about each written message
2249 Otherwise it will never try to send us the next one: it tries to keep
2250 exactly one message in-flight all the time.
2251 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
2252 in the client sink we always write data out synchronously.
2254 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
2256 * gst/rtsp-server/rtsp-stream.c:
2257 rtsp_server: Free thread pool before clean transport cache
2258 If not waiting for free thread pool before clean transport caches, there
2259 can be a crash if a thread is executing in transport list loop in
2260 function send_tcp_message.
2261 Also add a check if priv->send_pool in on_message_sent to avoid that a
2262 new thread is pushed during wait of free thread pool. This is possible
2263 since when waiting for free thread pool mutex have to be unlocked.
2265 === release 1.15.90 ===
2267 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
2273 * gst-rtsp-server.doap:
2277 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
2279 * gst/rtsp-server/rtsp-stream.c:
2280 rtsp-stream: Add support for GCM (RFC 7714)
2283 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2285 * gst/rtsp-server/rtsp-session-pool.c:
2286 session pool: fix missing klass-> in klass->create_session
2288 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2291 g-i: pass --quiet to g-ir-scanner
2292 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2293 that we get even if everything works just fine.
2294 We still get g-ir-scanner warnings and compiler warnings if
2295 we pass this option.
2297 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2300 g-i: silence 'nested extern' compiler warnings when building scanner binary
2301 We need a nested extern in our init section for the scanner binary
2302 so we can call gst_init to make sure GStreamer types are initialised
2303 (they are not all lazy init via get_type functions, but some are in
2304 exported variables). There doesn't seem to be any other mechanism to
2305 achieve this, so just remove that warning, it's not important at all.
2307 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2310 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2312 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2314 * gst/rtsp-server/rtsp-media.c:
2315 * tests/check/gst/media.c:
2316 rtsp-media: Handle set state when preparing.
2317 Handle the situation when a call to gst_rtsp_media_set_state is done
2318 when media status is preparing.
2319 Also add unit test for this scenario.
2320 The unit test simulate on a media level when two clients share a (live)
2322 Both clients have done SETUP and got responses. Now client 1 is doing
2323 play and client 2 is just closing the connection.
2324 Then without patch there are a problem when
2325 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2326 And client2 is doing closing connection we can end up in a call
2327 to gst_rtsp_media_set_state when
2328 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2329 shut down media is jumped over .
2330 With this patch and this scenario we wait until
2331 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2332 execute after that and now we will execute the logic for
2335 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2343 === release 1.15.2 ===
2345 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2351 * gst-rtsp-server.doap:
2355 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2357 * gst/rtsp-server/rtsp-media.c:
2358 * tests/check/gst/client.c:
2359 rtsp-media: Fix multicast use case with common media
2368 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2370 * gst/rtsp-server/rtsp-client.c:
2371 * gst/rtsp-server/rtsp-stream.c:
2372 * gst/rtsp-server/rtsp-stream.h:
2373 rtsp-server: remove recursive behavior
2374 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2376 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2378 * gst/rtsp-server/rtsp-client.c:
2379 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2380 And route all messages through the send_func if no send_messages_func
2382 We otherwise break backwards compatibility.
2384 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2386 * docs/libs/gst-rtsp-server-sections.txt:
2387 * gst/rtsp-server/rtsp-client.c:
2388 * gst/rtsp-server/rtsp-client.h:
2389 * gst/rtsp-server/rtsp-stream.c:
2390 rtsp-client: Add support for sending buffer lists directly
2391 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2393 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2395 * docs/libs/gst-rtsp-server-sections.txt:
2396 * gst/rtsp-server/rtsp-client.c:
2397 * gst/rtsp-server/rtsp-media.c:
2398 * gst/rtsp-server/rtsp-stream-transport.c:
2399 * gst/rtsp-server/rtsp-stream-transport.h:
2400 * gst/rtsp-server/rtsp-stream.c:
2401 * gst/rtsp-sink/gstrtspclientsink.c:
2402 rtsp-server: Add support for buffer lists
2403 This adds new functions for passing buffer lists through the different
2404 layers without breaking API/ABI, and enables the appsink to actually
2405 provide buffer lists.
2406 This should already reduce CPU usage and potentially context switches a
2407 bit by passing a whole buffer list from the appsink instead of
2408 individual buffers. As a next step it would be necessary to
2409 a) Add support for a vector of data for the GstRTSPMessage body
2410 b) Add support for sending multiple messages at once to the
2411 GstRTSPWatch and let it be handled internally
2412 c) Adding API to GOutputStream that works like writev()
2413 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2415 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2417 * gst/rtsp-server/rtsp-client.c:
2418 client: Fix crash in close handler
2419 The close handler could trigger a crash because it invalidated the
2420 watch_context while still leaving a source attached to it which would be
2421 cleaned up at a later point.
2423 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2425 * gst/rtsp-server/rtsp-stream.c:
2426 rtsp-stream: Use cached address when allocating sockets
2427 If an address/port was previously decided upon (ex: multicast in the
2428 SDP), then use that instead of re-creating another one
2429 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2431 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2433 * gst/rtsp-server/rtsp-media.c:
2434 rtsp-media: Fix race codition in finish_unprepare
2435 The previous fix for race condition around finish_unprepare where the
2436 function could be called twice assumed that the status wouldn't change
2437 during execution of the function. This assumption is incorrect as the
2438 state may change, for example if an error message arrives from the
2440 Instead a flag keeping track on whether the finish_unprepare function
2441 is currently executing is introduced and checked.
2442 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2444 === release 1.15.1 ===
2446 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2452 * gst-rtsp-server.doap:
2456 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2458 * gst/rtsp-server/rtsp-stream.c:
2459 Add source elements to the pipeline before activation
2460 In plug_src we changed the element state before adding it to
2461 the owner container. This prevented the pipeline from intercepting
2462 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2463 to assign a custom task pool.
2464 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2466 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2469 Automatic update of common submodule
2470 From ed78bee to 59cb678
2472 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2474 * examples/test-appsrc.c:
2475 examples: test-appsrc: fix coding style error
2477 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2479 * examples/test-appsrc.c:
2480 examples: test-appsrc: fix buffer leak
2482 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2484 * gst/rtsp-server/rtsp-media.c:
2485 rtsp-media: Update priv->blocked when linked streams are unblocked.
2486 Media is considered to be blocked when all streams that belong to
2487 that media are blocked.
2488 This patch solves the problem of inconsistent updates of
2489 priv->blocked that are not synchronized with the media state.
2491 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2493 * gst/rtsp-server/rtsp-media.c:
2494 rtsp-media: Don't block streams before seeking
2495 Before the seek operation is performed on media, it's required that
2496 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2497 At this stage, all transport parts (transport sinks) have been successfully
2498 added to the pipeline and there is no need for blocking the streams.
2500 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2502 * tests/check/gst/rtspserver.c:
2503 tests: rtspserver: Add shared media test case for TCP
2505 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2507 * gst/rtsp-server/rtsp-stream.c:
2508 rtsp-stream: Use seqnum-offset for rtpinfo
2509 The sequence number in the rtpinfo is supposed to be the first RTP
2510 sequence number. The "seqnum" property on a payloader is supposed to be
2511 the number from the last processed RTP packet. The sequence number for
2512 payloaders that inherit gstrtpbasepayload will not be correct in case of
2513 buffer lists. In order to fix the seqnum property on the payloaders
2514 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2515 "seqnum-offset" from the "stats" property contains the value of the
2516 very first RTP packet in a stream. The server will, however, try to look
2517 at the last simple in the sink element and only use properties on the
2518 payloader in case there no sink elements yet, and by looking at the last
2519 sample of the sink gives the server full control of which RTP packet it
2520 looks at. If the payloader does not have the "stats" property, "seqnum"
2521 is still used since "seqnum-offset" is only present in as part of
2522 "stats" and this is still an issue not solved with this patch.
2523 Needed for gst-plugins-base!17
2525 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2527 * gst/rtsp-server/rtsp-stream.c:
2528 rtsp-stream: Plug memory leak
2529 Attaching a GSource to a context will increase the refcount. The idle
2530 source will never be free'd since the initial reference is never
2533 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2536 Add Gitlab CI configuration
2537 This commit adds a .gitlab-ci.yml file, which uses a feature
2538 to fetch the config from a centralized repository. The intent is
2539 to have all the gstreamer modules use the same configuration.
2540 The configuration is currently hosted at the gst-ci repository
2541 under the gitlab/ci_template.yml path.
2542 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2544 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2547 * gst-rtsp-server.doap:
2548 Update git locations to gitlab
2550 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2552 * gst/rtsp-server/meson.build:
2553 meson: add new onvif types
2555 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2557 * gst/rtsp-server/meson.build:
2558 Add ONVIF subclass headers to the installed headers in meson.build too
2560 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2562 * gst/rtsp-server/rtsp-server-object.h:
2563 * gst/rtsp-server/rtsp-server.h:
2564 rtsp-server: Declare GstRTSPServer struct before anything else
2565 It's needed by all kinds of other headers, including the ones that are
2566 required for defining the GstRTSPServer struct itself and its API.
2568 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2570 * gst/rtsp-server/rtsp-onvif-client.h:
2571 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2572 * gst/rtsp-server/rtsp-onvif-media.h:
2573 * gst/rtsp-server/rtsp-onvif-server.h:
2574 Mark all ONVIF-specific subclasses as Since 1.14
2576 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2578 * gst/rtsp-server/Makefile.am:
2579 * gst/rtsp-server/meson.build:
2580 * gst/rtsp-server/rtsp-context.h:
2581 * gst/rtsp-server/rtsp-onvif-server.c:
2582 * gst/rtsp-server/rtsp-onvif-server.h:
2583 * gst/rtsp-server/rtsp-server-object.h:
2584 * gst/rtsp-server/rtsp-server-prelude.h:
2585 * gst/rtsp-server/rtsp-server.c:
2586 * gst/rtsp-server/rtsp-server.h:
2587 * gst/rtsp-server/rtsp-session.h:
2588 Include ONVIF types from single-include rtsp-server.h
2589 ... by actually making it a single-include header and moving everything
2590 related to the GstRTSPServer type to rtsp-server-object.h instead.
2591 Otherwise there are too many circular includes.
2592 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2594 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2596 * gst/rtsp-server/rtsp-client.c:
2597 * gst/rtsp-server/rtsp-latency-bin.c:
2598 * gst/rtsp-server/rtsp-stream.c:
2599 * gst/rtsp-server/rtsp-stream.h:
2600 rtsp-stream: use idle source in on_message_sent
2601 When the underlying layers are running on_message_sent, this sometimes
2602 causes the underlying layer to send more data, which will cause the
2603 underlying layer to run callback on_message_sent again. This can go on
2605 To break this chain, we introduce an idle source that takes care of
2606 sending data if there are more to send when running callback
2607 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2609 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2611 * gst/rtsp-server/rtsp-client.c:
2612 rtsp-client: Remove timeout GSource on cleanup
2613 Avoids ending up with races where a timeout would still be around
2614 *after* a client was gone. This could happen rather easily in
2615 RTSP-over-HTTP mode on a local connection, where each RTSP message
2616 would be sent as a different HTTP connection with the same tunnelid.
2617 If not properly removed, that timeout would then try to free again
2618 a client (and its contents).
2620 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2622 * gst/rtsp-server/Makefile.am:
2623 autotools: fix distcheck
2625 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2627 * gst/rtsp-server/Makefile.am:
2628 * gst/rtsp-server/meson.build:
2629 * gst/rtsp-server/rtsp-latency-bin.c:
2630 * gst/rtsp-server/rtsp-latency-bin.h:
2631 * gst/rtsp-server/rtsp-onvif-media.c:
2632 onvif: encapsulate onvif part into a bin
2633 ...and thus do not let onvif affect pipelines latency
2634 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2636 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2638 * tests/check/gst/client.c:
2639 tests: client: Avoid bind() failures in tests
2640 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2642 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2644 * gst/rtsp-server/rtsp-media-factory.c:
2645 * gst/rtsp-server/rtsp-media-factory.h:
2646 * gst/rtsp-server/rtsp-media.c:
2647 * gst/rtsp-server/rtsp-media.h:
2648 * gst/rtsp-server/rtsp-stream.c:
2649 * gst/rtsp-server/rtsp-stream.h:
2650 * tests/check/gst/client.c:
2651 * tests/check/gst/mediafactory.c:
2652 New property for socket binding to mcast addresses
2653 By default the multicast sockets are bound to INADDR_ANY,
2654 as it's not allowed to bind sockets to multicast addresses
2655 in Windows. This default behaviour can be changed by setting
2656 bind-mcast-address property on the media-factory object.
2657 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2659 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2662 * gst/rtsp-server/Makefile.am:
2663 * gst/rtsp-server/meson.build:
2664 * gst/rtsp-server/rtsp-address-pool.c:
2665 * gst/rtsp-server/rtsp-auth.c:
2666 * gst/rtsp-server/rtsp-client.c:
2667 * gst/rtsp-server/rtsp-context.c:
2668 * gst/rtsp-server/rtsp-media-factory-uri.c:
2669 * gst/rtsp-server/rtsp-media-factory.c:
2670 * gst/rtsp-server/rtsp-media.c:
2671 * gst/rtsp-server/rtsp-mount-points.c:
2672 * gst/rtsp-server/rtsp-params.c:
2673 * gst/rtsp-server/rtsp-permissions.c:
2674 * gst/rtsp-server/rtsp-sdp.c:
2675 * gst/rtsp-server/rtsp-server-prelude.h:
2676 * gst/rtsp-server/rtsp-server.c:
2677 * gst/rtsp-server/rtsp-session-media.c:
2678 * gst/rtsp-server/rtsp-session-pool.c:
2679 * gst/rtsp-server/rtsp-session.c:
2680 * gst/rtsp-server/rtsp-stream-transport.c:
2681 * gst/rtsp-server/rtsp-stream.c:
2682 * gst/rtsp-server/rtsp-thread-pool.c:
2683 * gst/rtsp-server/rtsp-token.c:
2685 libs: fix API export/import and 'inconsistent linkage' on MSVC
2686 Export rtsp-server library API in headers when we're building the
2687 library itself, otherwise import the API from the headers.
2688 This fixes linker warnings on Windows when building with MSVC.
2689 Fix up some missing config.h includes when building the lib which
2690 is needed to get the export api define from config.h
2691 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2693 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2695 * gst/rtsp-server/rtsp-media-factory.c:
2696 rtsp-media-factory: Add missing break statements
2697 This resulted in warnings/assertions whenever one accessed the
2698 max-mcast-ttl property.
2702 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2705 * meson_options.txt:
2706 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2708 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2711 * meson_options.txt:
2712 * tests/check/meson.build:
2713 meson: add option to disable build of rtspclientsink plugin
2715 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2717 * meson_options.txt:
2718 meson: re-arrange options
2720 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2723 * meson_options.txt:
2724 * tests/check/meson.build:
2725 * tests/meson.build:
2726 meson: Use feature option for tests option
2727 This was somehow missed the last time around.
2729 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2731 * gst/rtsp-server/meson.build:
2733 meson: Maintain macOS ABI through dylib versioning
2734 Requires Meson 0.48, but the feature will be ignored on older versions
2735 so it's safe to add it without bumping the requirement.
2737 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2739 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2741 * gst/rtsp-sink/meson.build:
2743 meson: add pkg-config file for the rtspclientsink plugin
2745 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2747 * gst/rtsp-server/rtsp-client.c:
2748 * tests/check/gst/client.c:
2749 rtsp-client: Avoid reuse of channel numbers for interleaved
2750 If a (strange) client would reuse interleaved channel numbers in
2751 multiple SETUP requests, we should not accept them. The channel
2752 numbers are used for looking up stream transports in the
2753 priv->transports hash table, and transports disappear from the table
2754 if channel numbers are reused.
2755 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2756 server to change the channel numbers suggested by the client.
2757 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2759 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2761 * tests/check/gst/client.c:
2762 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2763 Allow regex for matching transport header against expected pattern.
2764 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2766 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2768 * tests/check/meson.build:
2769 meson: There is no gstreamer-plugins-good-1.0.pc
2770 There is no installed version of that, only an uninstalled version.
2772 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2774 * gst/rtsp-server/rtsp-client.c:
2775 * tests/check/gst/stream.c:
2776 Fix indentation again
2778 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2780 * gst/rtsp-server/rtsp-client.c:
2781 * gst/rtsp-server/rtsp-stream.c:
2782 * gst/rtsp-server/rtsp-stream.h:
2783 * tests/check/gst/client.c:
2784 * tests/check/gst/stream.c:
2785 stream: Added a list of multicast client addresses
2786 When media is shared, the same media stream can be sent
2787 to multiple multicast groups. Currently, there is no API
2788 to retrieve multicast addresses from the stream.
2789 When calling gst_rtsp_stream_get_multicast_address() function,
2790 only the first multicast address is returned.
2791 With this patch, each multicast destination requested in SETUP
2792 will be stored in an internal list (call to
2793 gst_rtsp_stream_add_multicast_client_address()).
2794 The list of multicast groups requested by the clients can be
2795 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2796 There still exist some problems with the current implementation
2797 in the multicast case:
2798 1) The receiving part is currently only configured with
2799 regard to the first multicast client (see
2800 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2801 2) Secondly, of security reasons, some constraints should be
2802 put on the requested multicast destinations (see
2803 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2804 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2805 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2807 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2809 * gst/rtsp-server/rtsp-client.c:
2810 * gst/rtsp-server/rtsp-stream.c:
2811 * gst/rtsp-server/rtsp-stream.h:
2812 * tests/check/gst/client.c:
2813 stream: Choose the maximum ttl value provided by multicast clients
2814 The maximum ttl value provided so far by the multicast clients
2815 will be chosen and reported in the response to the current
2817 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2818 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2820 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2822 * gst/rtsp-server/rtsp-stream.c:
2823 * tests/check/gst/client.c:
2824 rtsp-stream: Don't require address pool in the transport specific case
2825 If "transport.client-settings" parameter is set to true, the client is
2826 allowed to specify destination, ports and ttl.
2827 There is no need for pre-configured address pool.
2828 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2829 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2831 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2833 * gst/rtsp-server/rtsp-client.c:
2834 * tests/check/gst/client.c:
2835 client: Don't reserve multicast address in the client setting case
2836 When two multicast clients request specific transport
2837 configurations, and "transport.client-settings" parameter is
2838 set to true, it's wrong to actually require that these two
2839 clients request the same multicast group.
2840 Removed test_client_multicast_invalid_transport_specific test
2841 cases as they wrongly require that the requested destination
2842 address is supposed to be present in the address pool, also in
2843 the case when "transport.client-settings" parameter is set to true.
2844 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2845 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2847 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2849 * gst/rtsp-server/rtsp-media-factory.c:
2850 * gst/rtsp-server/rtsp-media-factory.h:
2851 * gst/rtsp-server/rtsp-media.c:
2852 * gst/rtsp-server/rtsp-media.h:
2853 * gst/rtsp-server/rtsp-stream.c:
2854 * gst/rtsp-server/rtsp-stream.h:
2855 * tests/check/gst/mediafactory.c:
2856 Add new API for setting/getting maximum multicast ttl value
2857 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2858 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2860 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2862 * gst/rtsp-server/rtsp-stream.c:
2863 rtsp-stream: avoid duplicating the first multicast client
2864 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2865 clients were dynamically added and removed to the multicast
2866 udp sinks, as such we should no longer add a first client in
2867 set_multicast_socket_for_udpsink
2868 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2870 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2872 * gst/rtsp-server/rtsp-stream.c:
2873 Revert "rtsp-stream: avoid duplicating the first multicast client"
2874 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2875 Commits where accidentially squashed together
2877 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2879 * gst/rtsp-server/rtsp-client.c:
2880 * gst/rtsp-server/rtsp-media-factory.c:
2881 * gst/rtsp-server/rtsp-media-factory.h:
2882 * gst/rtsp-server/rtsp-media.c:
2883 * gst/rtsp-server/rtsp-media.h:
2884 * gst/rtsp-server/rtsp-stream.c:
2885 * gst/rtsp-server/rtsp-stream.h:
2886 * tests/check/gst/client.c:
2887 * tests/check/gst/mediafactory.c:
2888 Revert "Add new API for setting/getting maximum multicast ttl value"
2889 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2890 Commits where accidentially squashed together
2892 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2894 * gst/rtsp-server/rtsp-stream.c:
2895 * tests/check/gst/client.c:
2896 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2897 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2898 Commits where accidentially squashed together
2900 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2902 * gst/rtsp-server/rtsp-client.c:
2903 * gst/rtsp-server/rtsp-stream.c:
2904 * gst/rtsp-server/rtsp-stream.h:
2905 * tests/check/gst/client.c:
2906 * tests/check/gst/stream.c:
2907 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2908 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2909 Commits where accidentially squashed together
2911 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2913 * examples/test-auth-digest.c:
2914 examples: Fix indentation
2916 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2918 * gst/rtsp-server/rtsp-client.c:
2919 * gst/rtsp-server/rtsp-stream.c:
2920 * gst/rtsp-server/rtsp-stream.h:
2921 * tests/check/gst/client.c:
2922 * tests/check/gst/stream.c:
2923 stream: Choose the maximum ttl value provided by multicast clients
2924 The maximum ttl value provided so far by the multicast clients
2925 will be chosen and reported in the response to the current
2927 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2929 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2931 * gst/rtsp-server/rtsp-stream.c:
2932 * tests/check/gst/client.c:
2933 rtsp-stream: Don't require address pool in the transport specific case
2934 If "transport.client-settings" parameter is set to true, the client is
2935 allowed to specify destination, ports and ttl.
2936 There is no need for pre-configured address pool.
2937 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2939 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2941 * gst/rtsp-server/rtsp-client.c:
2942 * gst/rtsp-server/rtsp-media-factory.c:
2943 * gst/rtsp-server/rtsp-media-factory.h:
2944 * gst/rtsp-server/rtsp-media.c:
2945 * gst/rtsp-server/rtsp-media.h:
2946 * gst/rtsp-server/rtsp-stream.c:
2947 * gst/rtsp-server/rtsp-stream.h:
2948 * tests/check/gst/client.c:
2949 * tests/check/gst/mediafactory.c:
2950 Add new API for setting/getting maximum multicast ttl value
2951 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2953 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2955 * gst/rtsp-server/rtsp-stream.c:
2956 rtsp-stream: avoid duplicating the first multicast client
2957 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2958 clients were dynamically added and removed to the multicast
2959 udp sinks, as such we should no longer add a first client in
2960 set_multicast_socket_for_udpsink
2961 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2963 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2965 * gst/rtsp-server/Makefile.am:
2966 rtsp-server: Add gstreamer-base gir dir in autotools
2968 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2970 * gst/rtsp-server/rtsp-client.c:
2971 * gst/rtsp-server/rtsp-stream.c:
2972 rtsp-client: always allocate both IPV4 and IPV6 sockets
2973 multiudpsink does not support setting the socket* properties
2974 after it has started, which meant that rtsp-server could no
2975 longer serve on both IPV4 and IPV6 sockets since the patches
2976 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2978 When first connecting an IPV6 client then an IPV4 client,
2979 multiudpsink fell back to using the IPV6 socket.
2980 When first connecting an IPV4 client, then an IPV6 client,
2981 multiudpsink errored out, released the IPV4 socket, then
2982 crashed when trying to send a message on NULL nevertheless,
2983 that is however a separate issue.
2984 This could probably be fixed by handling the setting of
2985 sockets in multiudpsink after it has started, that will
2986 however be a much more significant effort.
2987 For now, this commit simply partially reverts the behaviour
2988 of rtsp-stream: it will continue to only create the udpsinks
2989 when needed, as was the case since the patches were merged,
2990 it will however when creating them, always allocate both
2991 sockets and set them on the sink before it starts, as was
2992 the case prior to the patches.
2993 Transport configuration will only error out if the allocation
2994 of UDP sockets fails for the actual client's family, this
2995 also downgrades the GST_ERRORs in alloc_ports_one_family
2996 to GST_WARNINGs, as failing to allocate is no longer
2998 https://bugzilla.gnome.org/show_bug.cgi?id=796875
3000 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3003 * meson_options.txt:
3004 meson: Convert common options to feature options
3005 These are necessary for gst-build to set options correctly. The
3006 remaining automagic option is cgroup support in examples.
3007 https://bugzilla.gnome.org/show_bug.cgi?id=795107
3009 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
3011 * gst/rtsp-server/rtsp-stream.c:
3012 rtsp-stream: Slightly simplify locking
3014 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
3016 * gst/rtsp-server/rtsp-client.c:
3017 * gst/rtsp-server/rtsp-stream-transport.c:
3018 * gst/rtsp-server/rtsp-stream-transport.h:
3019 * gst/rtsp-server/rtsp-stream.c:
3020 Limit queued TCP data messages to one per stream
3021 Before, the watch backlog size in GstRTSPClient was changed
3022 dynamically between unlimited and a fixed size, trying to avoid both
3023 unlimited memory usage and deadlocks while waiting for place in the
3024 queue. (Some of the deadlocks were described in a long comment in
3026 In the previous commit, we changed to a fixed backlog size of 100.
3027 This is possible, because we now handle RTP/RTCP data messages differently
3028 from RTSP request/response messages.
3029 The data messages are messages tunneled over TCP. We allow at most one
3030 queued data message per stream in GstRTSPClient at a time, and
3031 successfully sent data messages are acked by sending a "message-sent"
3032 callback from the GstStreamTransport. Until that ack comes, the
3033 GstRTSPStream does not call pull_sample() on its appsink, and
3034 therefore the streaming thread in the pipeline will not be blocked
3035 inside GstRTSPClient, waiting for a place in the queue.
3036 pull_sample() is called when we have both an ack and a "new-sample"
3037 signal from the appsink. Then, we know there is a buffer to write.
3038 RTSP request/response messages are not acked in the same way as data
3039 messages. The rest of the 100 places in the queue are used for
3040 them. If the queue becomes full of request/response messages, we
3041 return an error and close the connection to the client.
3042 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
3044 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
3046 * gst/rtsp-server/rtsp-client.c:
3047 rtsp-client: Use fixed backlog size
3048 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
3049 Preparation for the next commit, which changes to a different way of
3050 avoiding both deadlocks and unlimited memory usage with the watch
3053 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3055 * gst/rtsp-server/rtsp-media.c:
3056 rtsp-media: unref clock (if set) when finalizing
3057 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3059 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3061 * docs/libs/gst-rtsp-server-sections.txt:
3062 rtsp-media: add gst_rtsp_media_*_set_clock to docs
3063 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3065 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
3067 * gst/rtsp-server/rtsp-media-factory.c:
3068 media-factory: unref old clock when setting new clock
3069 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3071 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
3073 * gst/rtsp-server/rtsp-media-factory.c:
3074 media-factory: unref clock in finalize
3075 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3077 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
3079 * gst/rtsp-server/rtsp-onvif-media.c:
3080 rtsp-onvif-media: fix g-ir-scanner warnings
3082 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3085 .gitignore: add another example binary
3087 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
3089 * examples/meson.build:
3090 meson: add new test-appsrc2 example to meson build
3092 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
3094 * examples/Makefile.am:
3095 examples: fix build of new test-appsrc2 example
3096 Need to link against libgstapp-1.0.
3098 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
3100 * examples/.gitignore:
3101 * examples/Makefile.am:
3102 * examples/test-appsrc2.c:
3103 examples: Add test-appsrc2
3104 Add an example of feeding both audio and video into an RTSP
3105 pipeline via appsrc.
3107 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
3109 * gst/rtsp-server/rtsp-client.c:
3110 client: Strip transport parts as whitespaces could be around commas
3111 https://bugzilla.gnome.org/show_bug.cgi?id=758428
3113 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
3115 * gst/rtsp-server/rtsp-stream.c:
3116 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
3117 Fix race when setting up source elements.
3118 Since we set the source element(s) to PLAYING state before hooking
3119 them up to the downstream funnel, it's possible for the source element
3120 to receive packets before we actually get to linking it to the funnel,
3121 in which case buffers would be pushed out on an unlinked pad, causing
3122 it to error out and stop receiving more data.
3123 We fix this by blocking the source's srcpad until we have linked it.
3124 https://bugzilla.gnome.org/show_bug.cgi?id=796160
3126 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
3128 * gst/rtsp-server/rtsp-stream.c:
3129 rtsp-stream: Fix mismatch between allowed and configured protocols
3130 https://bugzilla.gnome.org/show_bug.cgi?id=796679
3132 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
3134 * gst/rtsp-server/rtsp-stream.c:
3135 rtsp-stream: Emit a signal when the SRTP decoder is created
3136 https://bugzilla.gnome.org/show_bug.cgi?id=778080
3138 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
3140 * gst/rtsp-server/rtsp-stream.c:
3141 rtsp-stream: Don't require presence of sinks in _get_*_socket()
3142 Transport specific sink elements are added to the pipeline
3143 in PLAY request and sockets are already created in SETUP so
3144 it's actually wrong to require the presence of sinks in
3145 _get_*_socket() functions.
3146 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3148 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
3150 * gst/rtsp-server/rtsp-stream.c:
3151 rtsp-stream: Update transport for multicast clients as well
3152 If a multicast client requests different transport settings
3153 than the existing one make sure that this new transport
3154 configuruation is propagated to the multicast udp sink.
3155 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3157 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
3159 * gst/rtsp-server/rtsp-stream.c:
3160 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
3161 And not on unicast udp sinks
3162 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3164 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
3166 * gst/rtsp-server/rtsp-address-pool.c:
3167 * gst/rtsp-server/rtsp-auth.c:
3168 * gst/rtsp-server/rtsp-client.c:
3169 * gst/rtsp-server/rtsp-media-factory-uri.c:
3170 * gst/rtsp-server/rtsp-media-factory.c:
3171 * gst/rtsp-server/rtsp-media.c:
3172 * gst/rtsp-server/rtsp-mount-points.c:
3173 * gst/rtsp-server/rtsp-server.c:
3174 * gst/rtsp-server/rtsp-session-media.c:
3175 * gst/rtsp-server/rtsp-session-pool.c:
3176 * gst/rtsp-server/rtsp-session.c:
3177 * gst/rtsp-server/rtsp-stream-transport.c:
3178 * gst/rtsp-server/rtsp-stream.c:
3179 * gst/rtsp-server/rtsp-thread-pool.c:
3180 Update for g_type_class_add_private() deprecation in recent GLib
3182 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
3184 * gst/rtsp-server/rtsp-auth.c:
3185 * gst/rtsp-server/rtsp-media.c:
3186 * gst/rtsp-server/rtsp-sdp.c:
3187 * gst/rtsp-server/rtsp-stream.c:
3190 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
3192 * examples/Makefile.am:
3193 * examples/test-video-disconnect.c:
3194 examples: Add test-video-disconnect example
3195 Simple example which cuts off all clients 10 seconds
3196 after the first one connects.
3198 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3200 * docs/libs/gst-rtsp-server-sections.txt:
3201 * examples/test-auth-digest.c:
3202 * gst/rtsp-server/rtsp-auth.c:
3203 * gst/rtsp-server/rtsp-auth.h:
3204 rtsp-auth: Add support for parsing .htdigest files
3205 Passwords are usually not stored in clear text, but instead
3206 stored already hashed in a .htdigest file.
3207 Add support for parsing such files, add API to allow setting
3208 a custom realm in RTSPAuth, and update the digest example.
3209 https://bugzilla.gnome.org/show_bug.cgi?id=796637
3211 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
3213 * gst/rtsp-sink/gstrtspclientsink.c:
3214 * gst/rtsp-sink/gstrtspclientsink.h:
3215 rtspclientsink: fix waiting for multiple streams
3216 We were previously only ever waiting for a single stream to notify it's
3217 blocked status through GstRTSPStreamBlocking. Actually count streams to
3219 Fixes rtspclientsink sending SDP's without out some of the input
3221 https://bugzilla.gnome.org/show_bug.cgi?id=796624
3223 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3225 * docs/libs/gst-rtsp-server-sections.txt:
3226 docs: add missing auth methods
3228 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3230 * gst/rtsp-server/rtsp-stream.c:
3231 rtsp-stream: only create funnel if it didn't exist already.
3232 This precented using multiple protocols for the same stream.
3233 https://bugzilla.gnome.org/show_bug.cgi?id=796634
3235 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3237 * examples/meson.build:
3238 meson: build auth-digest example
3240 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
3242 * gst/rtsp-server/rtsp-client.c:
3243 * gst/rtsp-server/rtsp-media.c:
3244 * gst/rtsp-server/rtsp-sdp.c:
3245 * gst/rtsp-server/rtsp-session-media.c:
3246 * gst/rtsp-server/rtsp-stream-transport.c:
3247 Get payloader stats only for the sending streams
3248 Get/set payloader properties only for streams that actually
3249 contain a payloader element.
3250 https://bugzilla.gnome.org/show_bug.cgi?id=796523
3252 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
3254 * gst/rtsp-server/Makefile.am:
3255 Makefile: Don't hardcode libtool for g-i build
3256 Similar to the other commits in core/base/bad
3258 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
3260 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3261 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
3262 https://bugzilla.gnome.org/show_bug.cgi?id=796229
3264 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
3266 * gst/rtsp-sink/gstrtspclientsink.c:
3267 rtspclientsink: Don't deadlock in preroll on early close
3268 If the connection is closed very early, the flushing
3269 marker might not get set and rtspclientsink can get
3270 deadlocked waiting for preroll forever.
3271 https://bugzilla.gnome.org/show_bug.cgi?id=786961
3273 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3276 * meson_options.txt:
3277 meson: Update option names to omit disable_ and with- prefixes
3278 Also yield common options to the outer project (gst-build in our case)
3279 so that they don't have to be set manually.
3281 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3284 meson: use -Wl,-Bsymbolic-functions where supported
3285 Just like the autotools build.
3287 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3290 * tests/check/Makefile.am:
3291 configure: check for -good and -bad plugins only in uninstalled setup
3292 Avoids confusing configure messages looking or a -good .pc file
3294 Also use plugindir variables that common macros set while at it.
3295 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3297 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3299 * gst/rtsp-server/rtsp-client.c:
3300 rtsp-client: Fix session timeout
3301 When streaming data over TCP then is not the keep-alive
3302 functionality working.
3303 The reason is that the function do_send_data have changed
3304 to boolean but the code is still checking the received result
3305 from send_func with GST_RTSP_OK.
3306 The result is that a successful send_func will always lead to
3307 that do_send_data is returning false and the keep-alive will
3309 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3311 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3313 * docs/libs/gst-rtsp-server-sections.txt:
3314 * gst/rtsp-server/rtsp-media.c:
3315 * gst/rtsp-server/rtsp-sdp.c:
3316 * gst/rtsp-server/rtsp-stream.c:
3317 * gst/rtsp-server/rtsp-stream.h:
3318 * gst/rtsp-sink/gstrtspclientsink.c:
3319 * gst/rtsp-sink/gstrtspclientsink.h:
3320 Implement support for ULP Forward Error Correction
3321 In this initial commit, interface is only exposed for RECORD,
3322 further work will be needed in rtspsrc to support this for
3324 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3326 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3328 * gst/rtsp-server/rtsp-onvif-media.c:
3329 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3330 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3331 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3332 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3333 the opposite, just like the ONVIF standard.
3334 Let's follow those RFCs as we're doing RTSP here, and add a property at
3335 a later time if needed to switch to the SDP RFC behaviour.
3336 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3338 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3341 Automatic update of common submodule
3342 From 3fa2c9e to ed78bee
3344 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3346 * gst/rtsp-server/rtsp-client.c:
3347 * gst/rtsp-server/rtsp-media-factory.c:
3348 * gst/rtsp-server/rtsp-media.c:
3349 * gst/rtsp-server/rtsp-stream.c:
3350 * tests/check/gst/rtspclientsink.c:
3351 gst: Run everything through gst-indent again
3353 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3355 * gst/rtsp-server/rtsp-media.c:
3356 * tests/check/gst/media.c:
3357 rtsp-media: query the position on active streams if media is complete
3358 If the media is complete, i.e. one or more streams have been configured
3359 with sinks, then we want to query the position on those streams only.
3360 A query on an incomplete stream may return a position that originates from
3362 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3364 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3366 * gst/rtsp-sink/gstrtspclientsink.c:
3367 rtspclientsink: make sure not to use freed string
3368 Set transport string to NULL after freeing it, so that
3369 at worst we get a NULL pointer if constructing a new
3370 transport string fails (which shouldn't really fail here).
3371 Also check return value of that, just in case.
3374 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3376 * gst/rtsp-server/rtsp-client.c:
3377 rtsp-client: do not free string passed to take_header
3379 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3381 * gst/rtsp-server/rtsp-stream.c:
3382 rtsp-stream: do not take lock in request_aux_receiver
3383 Added it right before pushing the previous commit, it is
3384 incorrect and deadlocks because this function gets called
3385 from the join_bin thread, which already holds the lock,
3386 that's the reason why request_aux_sender didn't take the
3389 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3391 * docs/libs/gst-rtsp-server-sections.txt:
3392 * gst/rtsp-server/rtsp-media-factory.c:
3393 * gst/rtsp-server/rtsp-media-factory.h:
3394 * gst/rtsp-server/rtsp-media.c:
3395 * gst/rtsp-server/rtsp-media.h:
3396 * gst/rtsp-server/rtsp-stream.c:
3397 * gst/rtsp-server/rtsp-stream.h:
3398 rtsp-server: add API to enable retransmission requests
3399 "do-retransmission" was previously set when rtx-time != 0,
3400 which made no sense as do-retransmission is used to enable
3401 the sending of retransmission requests, where as rtx-time
3402 is used by the peer to enable storing of buffers in order
3403 to respond to retransmission requests.
3404 rtsp-media now also provides a callback for the
3405 request-aux-receiver signal.
3406 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3408 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3410 * gst/rtsp-sink/gstrtspclientsink.c:
3411 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3412 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3414 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3416 * gst/rtsp-sink/gstrtspclientsink.c:
3417 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3418 This in order to be able to decrypt the RTCP backchannel
3419 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3421 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3423 * gst/rtsp-server/rtsp-client.c:
3424 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3425 When sending back an encrypted RTCP back channel, it is useful
3426 for the client to know the encryption key.
3427 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3429 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3431 * gst/rtsp-server/rtsp-client.c:
3432 * gst/rtsp-server/rtsp-stream.c:
3433 * gst/rtsp-server/rtsp-stream.h:
3434 rtsp-stream: extract handle_keymgmt from rtsp-client
3435 rtspclientsink will also need to parse KeyMgmt headers
3436 sent by the server to decrypt the RTCP backchannel stream
3437 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3439 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3441 * gst/rtsp-sink/gstrtspclientsink.c:
3442 * tests/check/gst/rtspclientsink.c:
3443 rtspclientsink: Fix client ports for the RTCP backchannel
3444 This was broken since the work for delayed transport creation
3445 was merged: the creation of the transports string depends on
3446 calling stream_get_server_port, which only starts returning
3447 something meaningful after a call to stream_allocate_udp_sockets
3448 has been made, this function expects a transport that we parse
3449 from the transport string ...
3450 Significant refactoring is in order, but does not look entirely
3451 trivial, for now we put a band aid on and create a second transport
3452 string after the stream has been completed, to pass it in
3453 the request headers instead of the previous, incomplete one.
3454 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3456 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3458 * gst/rtsp-server/rtsp-client.c:
3459 rtsp-client:Error handling when equal http session cookie
3460 There are some clients that are sending same session cookie on random
3462 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3464 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3466 * gst/rtsp-server/rtsp-media-factory-uri.c:
3467 rtsp-media-factory-uri: Fix compilation with latest GLib
3468 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3469 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3470 data->factory = g_object_ref (factory);
3473 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3481 === release 1.14.0 ===
3483 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3489 * gst-rtsp-server.doap:
3493 === release 1.13.91 ===
3495 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3501 * gst-rtsp-server.doap:
3505 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3507 * gst/rtsp-server/Makefile.am:
3508 * gst/rtsp-server/meson.build:
3509 * gst/rtsp-server/rtsp-address-pool.h:
3510 * gst/rtsp-server/rtsp-auth.h:
3511 * gst/rtsp-server/rtsp-client.h:
3512 * gst/rtsp-server/rtsp-context.h:
3513 * gst/rtsp-server/rtsp-media-factory-uri.h:
3514 * gst/rtsp-server/rtsp-media-factory.h:
3515 * gst/rtsp-server/rtsp-media.h:
3516 * gst/rtsp-server/rtsp-mount-points.h:
3517 * gst/rtsp-server/rtsp-onvif-client.h:
3518 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3519 * gst/rtsp-server/rtsp-onvif-media.h:
3520 * gst/rtsp-server/rtsp-onvif-server.h:
3521 * gst/rtsp-server/rtsp-params.h:
3522 * gst/rtsp-server/rtsp-permissions.h:
3523 * gst/rtsp-server/rtsp-sdp.h:
3524 * gst/rtsp-server/rtsp-server-prelude.h:
3525 * gst/rtsp-server/rtsp-server.h:
3526 * gst/rtsp-server/rtsp-session-media.h:
3527 * gst/rtsp-server/rtsp-session-pool.h:
3528 * gst/rtsp-server/rtsp-session.h:
3529 * gst/rtsp-server/rtsp-stream-transport.h:
3530 * gst/rtsp-server/rtsp-stream.h:
3531 * gst/rtsp-server/rtsp-thread-pool.h:
3532 * gst/rtsp-server/rtsp-token.h:
3533 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3534 We need different export decorators for the different libs.
3535 For now no actual change though, just rename before the release,
3536 and add prelude headers to define the new decorator to GST_EXPORT.
3538 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3540 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3541 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3542 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3544 === release 1.13.90 ===
3546 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3552 * gst-rtsp-server.doap:
3556 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3558 * gst/rtsp-server/rtsp-media-factory.c:
3559 * gst/rtsp-server/rtsp-permissions.c:
3560 permissions: add Since tags and example for new API
3562 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3564 * docs/libs/gst-rtsp-server-sections.txt:
3565 * gst/rtsp-server/rtsp-media-factory.c:
3566 * gst/rtsp-server/rtsp-media-factory.h:
3567 * gst/rtsp-server/rtsp-permissions.c:
3568 * gst/rtsp-server/rtsp-permissions.h:
3569 * tests/check/gst/permissions.c:
3570 permissions: more bindings-friendly API
3571 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3573 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3576 meson: enable more warnings
3578 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3580 * gst/rtsp-server/rtsp-client.c:
3581 rtsp-client: Place netaddress meta on packets received via TCP
3582 This allows us to later map signals from rtpbin/rtpsource back to the
3583 corresponding stream transport, and allows to do keep-alive based on
3584 RTCP packets in case of TCP media transport.
3585 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3587 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3589 * gst/rtsp-sink/gstrtspclientsink.c:
3590 rtspclientsink: if OPEN failed, unqueue next command
3591 As READY_TO_PAUSED can no longer return async, the RECORD
3592 command will be queued before the OPEN command fails
3593 (for example in case the server could not be connected),
3594 and record then waits for ever.
3595 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3597 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3599 * gst/rtsp-sink/gstrtspclientsink.c:
3600 rtspclientsink: fix retrieval of custom payloader caps
3601 If a bin is passed as the custom payloader, the caps of
3602 its factory will be empty, the correct way to obtain the caps
3603 is to query its sinkpad.
3605 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3607 * gst/rtsp-sink/gstrtspclientsink.c:
3608 rtspclientsink: fix extra unref of custom payloader
3610 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3612 * gst/rtsp-sink/gstrtspclientsink.c:
3613 rspclientsink: fix recent code indentation
3615 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3617 * gst/rtsp-sink/gstrtspclientsink.c:
3618 rtspclientsink: add missing get_type prototype
3620 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3622 * gst/rtsp-sink/gstrtspclientsink.c:
3623 rtspclientsink: allow setting payloader as pad property
3624 This was a FIXME item, and can be quite useful, also
3625 allowing to specify payloader properties from the command
3626 line, which is always nice.
3627 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3629 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3631 * gst/rtsp-server/rtsp-media.c:
3632 rtsp-media: Replace g_print() log line
3633 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3635 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3637 * gst/rtsp-server/rtsp-media.c:
3638 * tests/check/gst/rtspclientsink.c:
3639 rtsp-media: fix RECORD getting stuck
3640 The test_record case was working because async=false had
3641 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3642 but that was incorrect, as it should not be needed.
3643 Removing async=false made the test fail as expected, this is
3644 fixed by not trying to preroll when preparing the media for
3645 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3646 and our peer will not start sending media until it has received
3647 a response to that request, and sent and received a response
3648 to RECORD as well, thus obviously preventing preroll.
3649 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3651 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3653 * gst/rtsp-server/rtsp-auth.c:
3654 rtsp-auth: fix set_tls_authentication_mode annotation
3656 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3658 * gst/rtsp-server/rtsp-onvif-media.c:
3659 rtp-server: remove redefined variable
3660 res is a boolean variable which is defined in the function scope and
3661 redefined, with no reason, in the loop scope. This patch removes the
3663 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3665 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3667 * gst/rtsp-server/rtsp-media.c:
3668 * gst/rtsp-server/rtsp-stream.c:
3669 * gst/rtsp-server/rtsp-stream.h:
3670 stream: Add functions for checking if stream is receiver or sender
3671 ...and replace all checks for RECORD in GstRTSPMedia which are really
3672 for "sender-only". This way the code becomes more generic and introducing
3673 support for onvif-backchannel later on will require no changes in
3676 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3678 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3679 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3680 onvif: Make requires_backchannel() public
3681 ...in order to let subclasses building the onvif part of the pipeline
3682 check whether backchannel shall be included or not.
3684 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3686 * gst/rtsp-server/rtsp-onvif-media.c:
3687 rtsp-server: Switch around sendonly/recvonly attributes
3688 They are wrong in the ONVIF streaming spec. The backchannel should be
3689 recvonly and the normal media should be sendonly: direction is always
3690 from the point of view of the SDP offerer (the server) according to
3693 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3695 * docs/libs/gst-rtsp-server-docs.sgml:
3696 * docs/libs/gst-rtsp-server-sections.txt:
3697 * examples/.gitignore:
3698 * examples/Makefile.am:
3699 * examples/test-onvif-backchannel.c:
3700 * gst/rtsp-server/Makefile.am:
3701 * gst/rtsp-server/rtsp-media.h:
3702 * gst/rtsp-server/rtsp-onvif-client.c:
3703 * gst/rtsp-server/rtsp-onvif-client.h:
3704 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3705 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3706 * gst/rtsp-server/rtsp-onvif-media.c:
3707 * gst/rtsp-server/rtsp-onvif-media.h:
3708 * gst/rtsp-server/rtsp-onvif-server.c:
3709 * gst/rtsp-server/rtsp-onvif-server.h:
3710 * gst/rtsp-server/rtsp-sdp.c:
3711 * gst/rtsp-server/rtsp-sdp.h:
3712 rtsp: Add support for ONVIF backchannel
3713 This adds a new RTSP server, client, media-factory and media subclass
3714 for handling the specifics of the backchannel. Ideally this later can be
3715 extended with other ONVIF specific features.
3717 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3719 * gst/rtsp-server/rtsp-media.c:
3720 rtsp-media: Add support for sending+receiving medias
3721 We need to add an appsrc/appsink in that case because otherwise the
3722 media bin will be a sink and a source for rtpbin, causing a pipeline
3724 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3726 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3732 === release 1.13.1 ===
3734 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3738 * gst-rtsp-server.doap:
3742 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3744 * gst/rtsp-server/rtsp-session-pool.c:
3745 session-pool: remove nullable return annotation
3746 create_watch can only return NULL from the API guards, no
3749 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3751 * gst/rtsp-server/rtsp-media-factory.c:
3752 * gst/rtsp-server/rtsp-media.c:
3753 set_clock functions: Add nullable annotations
3755 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3757 * gst/rtsp-server/rtsp-auth.c:
3758 * gst/rtsp-server/rtsp-client.c:
3759 * gst/rtsp-server/rtsp-media-factory.c:
3760 * gst/rtsp-server/rtsp-media.c:
3761 * gst/rtsp-server/rtsp-mount-points.c:
3762 * gst/rtsp-server/rtsp-server.c:
3763 * gst/rtsp-server/rtsp-session-media.c:
3764 * gst/rtsp-server/rtsp-session-pool.c:
3765 * gst/rtsp-server/rtsp-session.c:
3766 * gst/rtsp-server/rtsp-stream-transport.c:
3767 * gst/rtsp-server/rtsp-stream.c:
3768 * gst/rtsp-server/rtsp-thread-pool.c:
3769 All around: add annotations and API guards
3771 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3773 * tests/test-cleanup.c:
3774 test-cleanup: bind any port
3775 The meson test suite runs tests in parallel, trying to bind
3776 a single port made the test fail.
3778 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3781 meson: make version numbers ints and fix int/string comparison
3782 WARNING: Trying to compare values of different types (str, int).
3783 The result of this is undefined and will become a hard error
3784 in a future Meson release.
3786 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3788 * gst/rtsp-server/rtsp-context.c:
3789 gst_rtsp_context_get_current: add (skip) annotation
3790 The return value type is defined with G_DEFINE_POINTER_TYPE,
3791 and gi emits the following warning:
3792 Invalid non-constant return of bare structure or union; register as
3793 boxed type or (skip)
3795 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3797 * gst/rtsp-server/rtsp-client.c:
3798 rtsp-client: add type annotations
3799 gi doesn't seem to be able to figure out the type of the
3800 signal parameters when defined with G_DEFINE_POINTER_TYPE
3802 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3805 autotools: use -fno-strict-aliasing where supported
3806 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3808 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3811 meson: use -fno-strict-aliasing where supported
3812 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3814 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3816 * gst/rtsp-server/rtsp-mount-points.c:
3817 mount-points: bail out of loop again when matching mount points
3818 Previous patch led to us iterating the entire sequence. Bail out
3819 of the loop again if we have a match but are moving away from it.
3820 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3822 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3824 * tests/check/gst/mountpoints.c:
3825 tests: mountpoints: add more checks for mount point path matching
3826 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3828 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3830 * gst/rtsp-server/rtsp-mount-points.c:
3831 mount-points: fix matching of paths where there's also an entry with a common prefix
3832 e.g. with the following mount points
3836 _match() would not match /raw/video and /raw/snapshot correctly.
3837 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3839 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3841 * docs/libs/gst-rtsp-server-sections.txt:
3842 * gst/rtsp-server/rtsp-permissions.c:
3843 * gst/rtsp-server/rtsp-permissions.h:
3844 * tests/check/gst/permissions.c:
3845 permissions: add some new API to make this usable from bindings
3846 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3848 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3850 * gst/rtsp-server/rtsp-token.c:
3851 rtsp-token: annotate constructors for bindings
3852 This maps _new_empty() to _new(), which also makes RTSPToken()
3853 work properly now. Since this API wasn't usable from bindings
3854 before, this should hopefully be fine.
3855 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3857 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3859 * docs/libs/gst-rtsp-server-sections.txt:
3860 * gst/rtsp-server/rtsp-token.c:
3861 * gst/rtsp-server/rtsp-token.h:
3862 * tests/check/gst/token.c:
3863 rtsp-token: add some API to set fields from bindings
3864 The existing functions are all vararg-based and as such
3865 not usable from bindings.
3866 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3868 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3870 * tests/check/gst/rtspclientsink.c:
3871 * tests/check/gst/rtspserver.c:
3872 * tests/check/gst/sessionpool.c:
3873 * tests/check/gst/stream.c:
3874 tests: fix indentation
3877 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3879 * tests/check/gst/rtspserver.c:
3880 tests: rtspserver: fix another ref leak
3881 Even if this didn't show up in valgrind.
3883 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3885 * tests/check/gst/rtspclientsink.c:
3886 tests: rtspclientsink: fix leak
3888 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3890 * tests/check/gst/rtspserver.c:
3891 test: rtspserver: plug memory leak in test_no_session_timeout
3892 In test_no_session_timeout, unref the rtsp session object when the
3894 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3896 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3898 * gst/rtsp-sink/gstrtspclientsink.c:
3899 rtpsclientsink: Initialize and clear newly added mutex and cond
3900 While it *did* work, glib would automatically create new mutex and cond
3901 ... which never got freed
3903 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3905 * gst/rtsp-server/rtsp-stream.c:
3906 rtsp-stream: Set multicast TTL on the multicast sockets
3907 And not if we do unicast UDP.
3908 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3910 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3912 * gst/rtsp-server/rtsp-stream.c:
3913 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3914 In the multicast case (as in test-multicast, not test-multicast2), the
3915 address could be allocated/reserved (and thus set) already without
3916 allocating the actual socket. We need to allocate the socket here still
3917 instead of just claiming that it was already allocated.
3918 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3920 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3922 * gst/rtsp-sink/gstrtspclientsink.c:
3923 * gst/rtsp-sink/gstrtspclientsink.h:
3924 rtspclientsink: Use the new rtsp-stream API
3925 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3927 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3929 * gst/rtsp-sink/gstrtspclientsink.c:
3930 * gst/rtsp-sink/gstrtspclientsink.h:
3931 rtspclientsink: Wait until OPEN has been scheduled
3932 Make sure that the sink thread has started opening connection
3933 to the server before continuing.
3934 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3936 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3939 Automatic update of common submodule
3940 From e8c7a71 to 3fa2c9e
3942 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3944 * gst/rtsp-server/rtsp-media.c:
3945 * gst/rtsp-server/rtsp-session-media.c:
3946 * gst/rtsp-server/rtsp-stream.c:
3947 rtsp-server: Minor doc fixes
3950 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3953 * tests/Makefile.am:
3954 tests: disable all tests when --disable-tests is used
3955 Move conditional subdir include into top level.
3956 Based on patch by: Joel Holdsworth
3957 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3959 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3962 * meson_options.txt:
3963 * tests/meson.build:
3964 meson: build more tests and add options to disable tests and examples
3966 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3968 * gst/rtsp-server/rtsp-session.c:
3969 Fix build when -Werror=deprecated-declarations is on
3970 As gst_rtsp_session_next_timeout is deprecated.
3972 ../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]
3973 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3975 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3976 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3977 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3980 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3983 Automatic update of common submodule
3984 From 3f4aa96 to e8c7a71
3986 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3988 * tests/check/gst/media.c:
3989 check/media: Add seekability test case: not all streams are active
3990 Media contains two streams but only one is complete and prepared
3992 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3994 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3996 * gst/rtsp-server/rtsp-stream.c:
3997 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3998 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4000 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4002 * gst/rtsp-server/rtsp-media.c:
4003 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
4004 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4006 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
4009 meson: remove vs_module_defs_dir variable which is no longer needed
4011 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
4013 * gst/rtsp-server/rtsp-session.h:
4016 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
4019 * gst/rtsp-server/meson.build:
4021 * win32/common/libgstrtspserver.def:
4022 win32: remove .def file with exports
4023 They're no longer needed, symbol exporting is now explicit
4024 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
4026 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4029 autotools: stop controlling symbol visibility with -export-symbols-regex
4030 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
4031 This should result in consistent behaviour for the autotools and
4034 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4036 * gst/rtsp-server/rtsp-media.h:
4037 * gst/rtsp-server/rtsp-server.h:
4038 * gst/rtsp-server/rtsp-session.c:
4039 * gst/rtsp-server/rtsp-session.h:
4040 rtsp-server: add missing GST_EXPORT and export deprecated funcs
4042 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
4044 * tests/check/gst/media.c:
4045 check: Add seekability testing on medias
4046 Make sure that once GstRTSPMedia are prepared they returned
4047 the expected seekability results
4048 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4050 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
4052 * docs/libs/gst-rtsp-server-sections.txt:
4053 * gst/rtsp-server/rtsp-media.c:
4054 * gst/rtsp-server/rtsp-stream.c:
4055 * gst/rtsp-server/rtsp-stream.h:
4056 * win32/common/libgstrtspserver.def:
4057 rtsp-media: Enable seeking query before pipeline is complete
4058 SDP are now provided *before* the pipeline is fully complete. In order
4059 to know whether a media is seekable or not therefore requires asking
4060 the invididual streams.
4061 API: gst_rtsp_stream_seekable
4062 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4064 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
4066 * gst/rtsp-server/rtsp-media.c:
4067 rtsp-media: Fix handling in default_unsuspend()
4068 Handle the case when streams are not blocked and media
4069 is suspended from PAUSED.
4070 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
4071 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4073 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
4075 * tests/check/gst/media.c:
4076 check/media: Fix thread pool leak.
4077 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
4078 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4080 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
4082 * gst/rtsp-server/rtsp-media.c:
4083 rtsp-media: Removed fakesink elements
4084 There is not need of adding fakesink elements to the media
4085 pipeline in the dynamic-payloader case.
4086 The media pipeline itself is dynamically updated with
4087 the receiver and sender parts that are based on the client
4088 transport information known after SETUP has been received.
4089 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
4090 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4092 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
4094 * gst/rtsp-server/rtsp-media.c:
4095 rtsp-media: Corrected ASYNC_DONE handling
4096 Media is complete when all the transport based parts are
4097 added to the media pipeline. At this point ASYNC_DONE is
4098 posted by the media pipeline and media is ready to enter
4100 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
4101 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4103 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
4105 * tests/check/gst/media.c:
4106 check/media: Check that prepared media can provide a SDP
4107 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
4109 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
4111 * gst/rtsp-server/rtsp-client.c:
4112 rtsp-client: Don't leak addr
4115 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
4117 * gst/rtsp-server/rtsp-client.c:
4118 * gst/rtsp-server/rtsp-session-media.c:
4119 * gst/rtsp-server/rtsp-stream.c:
4122 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
4124 * gst/rtsp-server/rtsp-media.c:
4125 rtsp-media: Don't unblock with remaining dynamic payloaders
4126 If we still have some dynamic paylaoders which haven't posted
4127 no-more-pads yet, don't go to PREPARED if one of the streams
4129 The risk was that we would end up not exposing/using all specified
4131 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
4132 then it will take a bit more time to start. But only if those 3
4133 conditions are present.
4134 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4136 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
4138 * gst/rtsp-server/rtsp-media.c:
4141 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
4143 * gst/rtsp-server/rtsp-media.c:
4144 rtsp-media: Don't set float on a gint64 variable
4145 Just use 0. Fixes 'undefined' behaviour from clang
4147 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
4149 * gst/rtsp-server/rtsp-media.c:
4150 rtsp-media: Fix previous commit
4151 We only want to count dynamic payloaders
4153 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
4155 * gst/rtsp-server/rtsp-media.c:
4156 * tests/check/gst/media.c:
4157 rtsp-media: Handle multiple dynamic elements
4158 If we have more than one dynamic payloader in the pipeline, we need
4159 to wait until the *last* one emits 'no-more-pads' before switching
4161 Failure to do so would result in a race where some of the streams
4162 wouldn't properly be prepared
4163 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4165 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4167 * win32/common/libgstrtspserver.def:
4168 win32: Fix exported symbols list
4170 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
4172 * gst/rtsp-server/rtsp-stream.c:
4173 rtsp-stream: Only update the RTP udpsink if it actually exists
4174 For send-only streams it does not exist, but the RTCP udpsink might.
4176 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
4178 * win32/common/libgstrtspserver.def:
4179 win32: Update exports
4181 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
4183 * gst/rtsp-server/rtsp-media.c:
4184 * gst/rtsp-server/rtsp-stream.c:
4185 * gst/rtsp-server/rtsp-stream.h:
4186 rtsp-media: seek on media pipelines that are complete
4187 Make sure that a seek is performed on pipelines that
4188 contain at least one sink element.
4189 Change-Id: Icf398e10add3191d104b1289de612412da326819
4190 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4192 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
4194 * gst/rtsp-server/rtsp-client.c:
4195 * gst/rtsp-server/rtsp-media.c:
4196 * gst/rtsp-server/rtsp-media.h:
4197 * gst/rtsp-server/rtsp-stream.c:
4198 * gst/rtsp-server/rtsp-stream.h:
4199 * tests/check/gst/client.c:
4200 * tests/check/gst/media.c:
4201 * tests/check/gst/rtspserver.c:
4202 * tests/check/gst/stream.c:
4203 Dynamically reconfigure pipeline in PLAY based on transports
4204 The initial pipeline does not contain specific transport
4205 elements. The receiver and the sender parts are added
4207 If the media is shared, the streams are dynamically
4208 reconfigured after each PLAY.
4209 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4211 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
4213 * gst/rtsp-server/rtsp-stream.c:
4214 rtsp-stream: obtain stream position from pad
4215 If no sinks have been added yet, obtain the current and
4216 the stop position of the stream from the send_src pad.
4217 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
4218 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4220 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
4222 * gst/rtsp-server/rtsp-session-media.c:
4223 * gst/rtsp-server/rtsp-session-media.h:
4224 rtsp-session-media: add function to get a list of transports
4225 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
4226 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4228 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
4230 * gst/rtsp-server/rtsp-stream.c:
4231 * gst/rtsp-server/rtsp-stream.h:
4232 rtsp-stream: add functions to get rtp and rtcp multicast sockets
4233 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
4234 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4236 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
4238 * gst/rtsp-server/rtsp-stream.c:
4239 stream: set async=sync=false only for RTCP appsink
4240 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
4241 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4243 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
4245 * gst/rtsp-server/rtsp-media.c:
4246 rtsp-media: return minimum value in query position case
4247 The minimum position should be returned as we are interested
4248 in the whole interval.
4249 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
4250 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4252 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
4254 * gst/rtsp-server/rtsp-session.c:
4255 * tests/check/gst/rtspserver.c:
4256 rtsp-session: Handle the case when timeout=0
4257 According to the documentation, a timeout of value 0 means
4258 that the session never timeouts. This adds handling of that.
4259 If timeout=0 we just return with a -1 from
4260 gst_rtsp_session_next_timeout_usec ().
4261 https://bugzilla.gnome.org/show_bug.cgi?id=785058
4263 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4265 * gst/rtsp-sink/gstrtspclientsink.c:
4266 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
4267 https://bugzilla.gnome.org/show_bug.cgi?id=785024
4269 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
4271 * docs/libs/gst-rtsp-server-sections.txt:
4272 * gst/rtsp-server/rtsp-media-factory.c:
4273 docs: add media factory transport mode accessors
4274 and fix the documentation for the return value of the getter
4276 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
4278 * gst/rtsp-server/rtsp-client.c:
4279 rtsp-client: unref 'pipelined_requests' in finalize
4280 The hash table priv->pipelined_requests is not unref:ed in the
4281 finalize funktion. Make sure it is.
4282 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4284 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4286 * gst/rtsp-server/rtsp-media.c:
4287 rtsp-media: Initialize scalar variable
4290 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4292 * win32/common/libgstrtspserver.def:
4293 win32: Update export file
4295 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4297 * gst/rtsp-server/rtsp-client.c:
4298 * gst/rtsp-server/rtsp-media.c:
4299 * gst/rtsp-server/rtsp-media.h:
4300 Start support for RTSP 2.0
4301 This adds basic support for new 2.0 features, though the protocol is
4302 subposdely backward incompatible, most semantics are the sames.
4305 * version negotiation
4306 * pipelined requests support
4307 * Media-Properties support
4308 * Accept-Ranges support
4310 * gst_rtsp_media_seekable
4311 The RTSP methods that have been removed when using 2.0 now return
4313 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4315 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4317 * gst/rtsp-server/rtsp-stream.c:
4318 stream: Use stream duration as stream-stop if segment was not configured with a stop
4319 Allowing client to know stream duration when no seeking happened.
4320 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4322 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4324 * gst/rtsp-server/rtsp-media-factory.c:
4325 rtsp-media-factory: Don't cache any media if NULL was returned as key
4326 The docs already mentioned this, but we actually stored it in the hash
4327 table with key==NULL and leaked its reference forever.
4329 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4331 * gst/rtsp-sink/gstrtspclientsink.c:
4332 * gst/rtsp-sink/gstrtspclientsink.h:
4333 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4334 This is a simple port of:
4335 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4336 * c438545dc9e2f14f657bc0ef261fff726449867b
4337 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4338 in gst-plugins-good.
4340 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4342 * gst/rtsp-server/rtsp-sdp.c:
4343 sdp: fix Memory leak in error case
4344 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4346 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4348 * pkgconfig/meson.build:
4349 meson: don't install -uninstalled.pc file
4350 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4352 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4355 Automatic update of common submodule
4356 From 48a5d85 to 3f4aa96
4358 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4360 * gst/rtsp-server/rtsp-client.c:
4361 rtsp-client: Fix typo in debug message
4363 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4366 meson: hide symbols by default unless explicitly exported
4368 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4370 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4371 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4372 Fixes meson warning about undefined @srcdir@.
4374 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4376 * tests/meson.build:
4377 meson: skip tests on windows for now
4378 As we do in the other modules. As libgstcheck is currently not
4379 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4380 the subproject 'gstreamer' does not exist"" Meson error.
4382 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4384 * gst/rtsp-server/rtsp-stream.c:
4385 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4386 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4387 multiudpsink's last-sample always comes from the payloader. Which
4388 is wrong if auxiliary streams are multiplexed in the same stream.
4389 So check the buffer's ssrc against the caps'ssrc before to use its
4390 seqnum. If not the same ssrc just use the payloader as done prior
4391 the commit above or when there is no last-sample yet.
4392 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4394 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4397 meson: Allow using glib as a subproject
4399 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4402 meson: fix with-package-name option
4403 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4405 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4408 Distribute meson_options.txt
4410 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4413 And config.h.meson is no longer dist either
4415 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4419 meson: config.h.meson is no longer needed
4421 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4423 * tests/check/meson.build:
4424 * tests/meson.build:
4425 meson: Fix building tests and activate them again
4427 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4429 * tests/check/meson.build:
4430 meson: Do not use path separator in test names
4431 Avoiding warnings like:
4432 WARNING: Target "elements/audioamplify" has a path separator in its name.
4434 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4437 * meson_options.txt:
4438 meson: add options to set package name and origin
4439 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4441 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4443 * gst/rtsp-server/rtsp-address-pool.h:
4444 * gst/rtsp-server/rtsp-auth.h:
4445 * gst/rtsp-server/rtsp-client.h:
4446 * gst/rtsp-server/rtsp-context.h:
4447 * gst/rtsp-server/rtsp-media-factory-uri.h:
4448 * gst/rtsp-server/rtsp-media-factory.h:
4449 * gst/rtsp-server/rtsp-media.h:
4450 * gst/rtsp-server/rtsp-mount-points.h:
4451 * gst/rtsp-server/rtsp-params.h:
4452 * gst/rtsp-server/rtsp-permissions.h:
4453 * gst/rtsp-server/rtsp-sdp.h:
4454 * gst/rtsp-server/rtsp-server.h:
4455 * gst/rtsp-server/rtsp-session-media.h:
4456 * gst/rtsp-server/rtsp-session-pool.h:
4457 * gst/rtsp-server/rtsp-session.h:
4458 * gst/rtsp-server/rtsp-stream-transport.h:
4459 * gst/rtsp-server/rtsp-stream.h:
4460 * gst/rtsp-server/rtsp-thread-pool.h:
4461 * gst/rtsp-server/rtsp-token.h:
4462 Mark symbols explicitly for export with GST_EXPORT
4464 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4467 * gst/rtsp-sink/Makefile.am:
4468 Remove plugin specific static build option
4469 Static and dynamic plugins now have the same interface. The standard
4470 --enable-static/--enable-shared toggle are sufficient.
4472 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4478 === release 1.12.0 ===
4480 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4486 * gst-rtsp-server.doap:
4490 === release 1.11.91 ===
4492 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4498 * gst-rtsp-server.doap:
4502 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4505 Automatic update of common submodule
4506 From 60aeef6 to 48a5d85
4508 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4510 * gst/rtsp-server/rtsp-media-factory.c:
4511 * gst/rtsp-server/rtsp-media.c:
4512 * gst/rtsp-server/rtsp-session.c:
4513 * gst/rtsp-server/rtsp-stream.c:
4514 gi: Fix some annotations and docstrings
4516 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4518 * gst/rtsp-server/meson.build:
4520 * meson_options.txt:
4523 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4527 Automatic update of common submodule
4528 From 39ac2f5 to 60aeef6
4530 === release 1.11.90 ===
4532 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4538 * gst-rtsp-server.doap:
4542 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4544 * examples/test-launch.c:
4545 examples: make test-launch pipeline shared by default as well
4547 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4549 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4550 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4551 Just the build dir is not going to work for srcdir!=builddir.
4553 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4556 meson: Update version
4558 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4563 === release 1.11.2 ===
4565 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4571 * gst-rtsp-server.doap:
4574 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4577 meson: dist meson build files
4578 Ship meson build files in tarballs, so people who use tarballs
4579 in their builds can start playing with meson already.
4581 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4583 * examples/test-record.c:
4584 examples/test-record: Add extra line to initial printout
4585 Add an example line of how to deliver a stream to the
4588 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4590 * gst/rtsp-server/rtsp-client.c:
4591 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4592 If there is no Content-Length header, no body would be allocated and the
4593 '\0' would also not be appended to the body.
4595 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4597 * gst/rtsp-server/rtsp-client.c:
4598 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4599 While they logically have 0 bytes length, GstRTSPConnection is appending
4600 a '\0' to everything making the size be 1 instead.
4602 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4607 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4609 * gst/rtsp-server/rtsp-session.c:
4610 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4611 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4614 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4619 === release 1.11.1 ===
4621 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4627 * gst-rtsp-server.doap:
4628 * win32/common/libgstrtspserver.def:
4631 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4633 * gst/rtsp-server/rtsp-stream.c:
4634 rtsp-stream: corrected if-statement in _get_server_port()
4635 This bug was accidentally introduced while fixing a segfault
4636 in _get_server_port() function.
4637 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4639 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4641 * gst/rtsp-server/rtsp-stream.c:
4642 * tests/check/gst/stream.c:
4643 rtsp-stream: fixed segmenation fault in _get_server_port()
4644 Calling function gst_rtsp_stream_get_server_port() results in
4645 segmenation fault in the RTP/RTSP/TCP case.
4646 Port that the server will use to receive RTCP makes only
4647 sense in the UDP case, however the function should handle
4648 the TCP case in a nicer way.
4649 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4651 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4653 * gst/rtsp-server/rtsp-media-factory.c:
4654 dosc: Fix a little typo
4655 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4657 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4659 * pkgconfig/Makefile.am:
4660 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4661 * pkgconfig/meson.build:
4662 meson: generate pkg-config -uninstalled pc files
4663 Generating those files is useful for users building the GStreamer stack
4664 using meson and having to link it to another project which is still
4665 using the autotools.
4666 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4668 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4670 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4671 pkgconfig: fix -uninstalled pc file
4672 pcfiledir was never defined so the paths were wrong.
4673 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4675 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4677 * gst/rtsp-server/rtsp-stream.c:
4678 * tests/check/gst/rtspserver.c:
4679 rtsp-stream: Fixed TCP transport case
4680 Make sure that the appsink element is actually added to
4681 the bin before trying to link it with the elements in it.
4682 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4684 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4690 Remove generated .spec file
4691 Likely extremely bitrotten, and we should not ship this anyway.
4693 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4696 Automatic update of common submodule
4697 From f980fd9 to 39ac2f5
4699 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4701 * gst/rtsp-server/rtsp-media.c:
4702 media: Fix pt map caps
4703 Since decryption is handled within rtpbin, all outcoming stream
4704 caps will be application/x-rtp (i.e. regular rtp)
4705 Fixes RECORD with SRTP streams
4707 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4709 * gst/rtsp-server/rtsp-media-factory.c:
4710 media-factory: Create media objects with the proper transport mode
4711 The function called immediately afterwards (collect_streams()) will
4712 need it to work properly
4714 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4716 * gst/rtsp-server/rtsp-auth.c:
4717 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4719 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4721 * gst/rtsp-server/rtsp-media-factory.c:
4722 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4723 We're going to put a pipeline into a pipeline otherwise, which is not
4726 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4728 * gst/rtsp-server/rtsp-media.c:
4729 media: Fix race condition around finish_unprepare() if called multiple time
4730 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4732 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4734 * gst/rtsp-sink/gstrtspclientsink.c:
4735 rtspclientsink: Don't leave stale pointer after unref
4736 Fix a warning on shutdown - don't keep a pointer to an
4737 alread-unreffed object.
4739 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4742 common: use https protocol for common submodule
4743 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4745 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4747 * gst/rtsp-server/rtsp-stream.c:
4748 stream: block the output of rtpbin instead of the source pipeline
4749 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4750 detection of the srtp rollover counter to add to the SDP.
4751 Unfortunately, it was incomplete for live pipelines where the logic
4752 blocks the source bin before creating the SDP and thus would never have
4753 the necessary informaiton to create a correct SDP with srtp encryption.
4754 Move the pad blocks to rtpbin's output pads instead so that the
4755 necessary information can be created before we need the information for
4757 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4759 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4761 * gst/rtsp-server/rtsp-client.c:
4762 rtsp-client: add IDLE timeout, before session exists
4763 The RTSP server will not timeout an idle RTSP connection
4764 (note this is different from doing timeout on a RTSP
4766 At least for Apache this is a problem when running RTSP over
4767 HTTPS since it uses one of the threads (there is a rather
4768 limited number) that are available for handling requests.
4769 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4771 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4776 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4778 * gst/rtsp-server/rtsp-stream.c:
4779 rtsp-stream: Set close-socket FALSE on UDP src:es
4780 With this RTSP server can use the sockets independent on the udpsrc
4782 When the udp src is finalized it will unref socket and when g_socket
4783 is finalized the socket will be closed.
4784 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4786 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4788 * gst/rtsp-sink/gstrtspclientsink.c:
4789 rtspclientsink: Move to new helper function to parse authentication responses
4790 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4792 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4794 * examples/Makefile.am:
4795 * examples/test-auth-digest.c:
4796 * gst/rtsp-server/rtsp-auth.c:
4797 * gst/rtsp-server/rtsp-auth.h:
4798 * win32/common/libgstrtspserver.def:
4799 rtsp-auth: Add support for Digest authentication
4800 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4802 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4805 * gst/rtsp-server/meson.build:
4807 * tests/check/meson.build:
4809 * win32/common/libgstrtspserver.def:
4810 Enable building with MSVC
4811 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4813 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4816 meson: gstreamer gst_check_dep does not exist on windows
4818 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4820 * gst/rtsp-server/rtsp-client.c:
4821 client: update do_send_message to match type GstRTSPClientSendFunc
4822 This type mismatch fails building with MSVC
4823 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4825 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4827 * gst/rtsp-server/rtsp-sdp.c:
4828 rtsp-sdp: Fix indentation
4830 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4832 * gst/rtsp-server/rtsp-media.c:
4833 rtsp-media: Only signal "new-state" if the state has actually changed
4834 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4836 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4838 * gst/rtsp-server/rtsp-client.c:
4839 * gst/rtsp-server/rtsp-client.h:
4840 client: emit signal in the beginning of each rtsp request
4841 These signals let the application validate the requests, configure the
4842 media/stream in a certain way and also generate error status code in
4843 case of error or bad request.
4844 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4846 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4849 meson: update version
4851 === release 1.11.0 ===
4853 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4858 === release 1.10.0 ===
4860 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4866 * gst-rtsp-server.doap:
4869 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4871 * tests/check/gst/rtspserver.c:
4872 * tests/check/gst/stream.c:
4873 tests: try to avoid using the same ports in different tests
4874 Causes problems with client multicast tests otherwise if
4875 tests are run in parallel.
4876 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4878 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4880 * tests/check/gst/client.c:
4881 tests: client: use fail_unless_equals_foo() for better failure reporting
4883 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4885 * gst/rtsp-server/rtsp-client.c:
4886 rtsp-client: Session filter in unwatch session
4887 Call session filter with filter_session_media as paramer in
4888 client_unwatch_session if using drop_backlog = FALSE.
4889 In client_unwatch_session its allowed to grow the watchs backlog.
4890 If using drop_backlog = FALSE and the backlog is full it will cause
4891 a deadlock when setting session media state to NULL
4892 if the backlog is not allowed to grow.
4893 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4895 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4898 meson: add fallbacks for gst modules
4901 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4903 * gst/rtsp-server/rtsp-client.c:
4904 rtsp-client: Fix factory leaking in find_media() in error cases
4905 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4907 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4909 * gst/rtsp-server/rtsp-stream.c:
4910 stream: Fix randomly missing streams from SDP with dynamic elements
4911 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4912 "pad-added" signal. In that case priv->srcpad could already have its caps,
4913 and they'll be sent to priv->send_src[0] pad. That means that when it
4914 connects "notify::caps" signal, that pad could already have received its
4915 caps and the signal won't be emitted anymore.
4916 In that case priv->caps stay to NULL and when building the SDP that stream
4917 gets ignored. Leading to missing video or audio when playing in client side.
4918 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4920 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4923 meson: update version
4925 === release 1.9.90 ===
4927 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4933 * gst-rtsp-server.doap:
4936 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4938 * gst/rtsp-server/rtsp-media-factory.c:
4939 * gst/rtsp-server/rtsp-media.c:
4940 * gst/rtsp-server/rtsp-stream.c:
4941 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4942 To prevent any possibly confusion with IPs or anything else.
4943 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4945 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4947 * gst/rtsp-server/rtsp-media-factory.c:
4948 * gst/rtsp-server/rtsp-media.c:
4949 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4950 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4952 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4955 configure: Depend on gstreamer 1.9.2.1
4957 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4961 Automatic update of common submodule
4962 From b18d820 to f980fd9
4964 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4968 Automatic update of common submodule
4969 From 6f2d209 to b18d820
4971 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4973 * gst/rtsp-server/rtsp-stream.c:
4974 rtsp-stream: Remove unused _locked() variant of a function
4975 It was added during refactoring.
4977 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4979 * gst/rtsp-server/rtsp-stream.c:
4980 stream: cosmetic cleanup
4981 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4983 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4985 * gst/rtsp-server/rtsp-stream.c:
4986 stream: Compare IP addresses case insensitive in more places
4987 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4989 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4992 * gst/rtsp-server/rtsp-stream.c:
4993 stream: Fix leaked joined_bin
4994 There is no need to keep a strong ref on it, and _leave_bin() was
4995 setting it to NULL before calling g_clear_object() so it was leaked.
4996 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4998 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5000 * gst/rtsp-server/rtsp-stream.c:
5001 rtsp-stream: Compare IP address strings case insensitive
5002 Otherwise IPv6 addresses might fail this comparision.
5004 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
5006 * gst/rtsp-server/rtsp-stream.c:
5007 rtsp-stream: Bind multicast sockets to ANY as before
5008 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
5010 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
5012 * gst/rtsp-server/rtsp-session.c:
5013 rtsp-session: Fix segfault when doing keep-alive after removing the session
5014 If keep-alive happens after removing the session but before finalizing the
5015 stream transport, we would segfault.
5016 https://bugzilla.gnome.org/show_bug.cgi?id=750544
5018 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
5020 * gst/rtsp-server/rtsp-stream.c:
5021 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
5022 Adding them later will cause deadlocks due to
5023 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
5024 2) adding the multicast sink
5025 3) waiting for it to get data to preroll again
5026 3) never happens because the queues after the tee are full.
5028 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
5030 * gst/rtsp-server/rtsp-stream.c:
5031 rtsp-stream: Fix up various multicast related issues
5033 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
5035 * tests/check/gst/stream.c:
5036 tests: Fix compilation
5038 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5040 * gst/rtsp-server/rtsp-client.c:
5041 * gst/rtsp-server/rtsp-stream.c:
5042 * tests/check/gst/stream.c:
5043 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
5044 This is basically reverting changes introduced in commit f62a9a7,
5045 because it was introducing various regressions:
5046 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
5047 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
5048 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
5049 - If a mcast client connects, it creates a new socket in SETUP to try to respect
5050 the destination/port given by the client in the transport, and overrides the
5051 socket already set on the udpsink element. That means that if we already had a
5052 client connected, the source address on the udp packets it receives suddenly
5054 - If a 2nd mcast client connects, the destination/port in its transport is
5055 ignored but its transport wasn't updated.
5056 What this patch does:
5057 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
5058 - Always have a tee+queue when udp is enabled. This could be optimized
5059 again in a later patch, but is more complicated. If no unicast clients
5060 connects then those elements are useless, this could be also optimized
5062 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
5063 seperated from those for unicast clients. Since we already support only
5064 one mcast address, we also create only one set of elements.
5065 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5067 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5069 * gst/rtsp-server/rtsp-stream.c:
5070 stream: factor our plug_src function
5071 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5073 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5075 * gst/rtsp-server/rtsp-stream.c:
5076 stream: factor out plug_sink function
5077 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5079 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5081 * gst/rtsp-server/rtsp-stream.c:
5082 stream: small documentation clarification
5083 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5085 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5087 * gst/rtsp-server/rtsp-stream.c:
5088 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
5089 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5091 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5093 * gst/rtsp-server/rtsp-stream.c:
5094 stream: Keep a ref on joined bin
5095 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5097 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5099 * gst/rtsp-server/rtsp-stream.c:
5100 stream: code cleanup
5101 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5103 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5105 * gst/rtsp-server/rtsp-stream.c:
5106 stream: small fix in error code path
5107 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5109 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5111 * gst/rtsp-server/rtsp-stream.c:
5112 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
5113 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
5114 but keeps unit tests.
5115 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5117 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
5122 === release 1.9.2 ===
5124 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
5130 * gst-rtsp-server.doap:
5133 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
5136 * examples/meson.build:
5138 * gst/rtsp-server/meson.build:
5139 * gst/rtsp-sink/meson.build:
5141 * pkgconfig/meson.build:
5142 * tests/check/meson.build:
5143 * tests/meson.build:
5144 Add support for Meson as alternative/parallel build system
5145 https://github.com/mesonbuild/meson
5147 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
5150 * tests/check/Makefile.am:
5151 build: silence error about pthread for 'make check' in osx
5152 Fixes "clang: error: argument unused during compilation: '-pthread'"
5154 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
5156 * gst/rtsp-server/rtsp-client.c:
5157 rtsp-client: Fix leaking of media in error cases
5158 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
5159 and myself to make the media refcounting a bit easier to follow.
5160 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5162 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
5164 * gst/rtsp-server/rtsp-client.c:
5165 rtsp-client: Fix leaking of session in error cases
5166 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5168 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
5171 Automatic update of common submodule
5172 From f363b32 to f49c55e
5174 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
5179 === release 1.9.1 ===
5181 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5187 * gst-rtsp-server.doap:
5190 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
5193 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
5194 https://bugzilla.gnome.org/show_bug.cgi?id=767463
5196 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5199 Automatic update of common submodule
5200 From ac2f647 to f363b32
5202 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5204 * gst/rtsp-server/rtsp-sdp.c:
5205 * gst/rtsp-server/rtsp-sdp.h:
5206 * gst/rtsp-server/rtsp-stream.c:
5207 * gst/rtsp-server/rtsp-stream.h:
5208 sdp: add rollover counters for all sender SSRC
5209 We add different crypto sessions in MIKEY, one for each sender
5210 SSRC. Currently, all of them will have the same security policy, 0.
5211 The rollover counters are obtained from the srtpenc element using the
5213 https://bugzilla.gnome.org/show_bug.cgi?id=730539
5215 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5217 * gst/rtsp-server/rtsp-media-factory.h:
5218 * gst/rtsp-server/rtsp-server.h:
5219 docs: fix some typos
5221 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
5223 * gst/rtsp-server/Makefile.am:
5224 g-i: pass compiler env to g-ir-scanner
5225 It's what introspection.mak does as well. Should
5226 fix spurious build failures on gnome-continuous
5227 (caused by g-ir-scanner getting compiler details
5228 via python which is broken in some environments
5229 so passing the compiler details bypasses that).
5231 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
5233 * gst/rtsp-server/rtsp-session.c:
5234 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
5235 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
5236 https://bugzilla.gnome.org/show_bug.cgi?id=766619
5238 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
5240 * gst/rtsp-sink/gstrtspclientsink.c:
5241 rtspclientsink: Check return value of sscanf
5242 And just make sure we always have 0/0 if we have an error
5245 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
5247 * gst/rtsp-server/rtsp-stream.c:
5248 * tests/check/gst/rtspserver.c:
5249 * tests/check/gst/stream.c:
5250 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
5251 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
5252 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
5253 - Create unit test for shared media.
5254 https://bugzilla.gnome.org/show_bug.cgi?id=764744
5256 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5258 * gst/rtsp-server/rtsp-stream.c:
5259 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
5260 For IPv6 addresses, binding to a multicast group does not work on Linux
5261 either. Always bind to ANY and then later join the multicast group.
5262 https://bugzilla.gnome.org/show_bug.cgi?id=764679
5264 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
5267 Automatic update of common submodule
5268 From 6f2d209 to ac2f647
5270 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
5272 * gst/rtsp-server/rtsp-thread-pool.c:
5273 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
5274 Clarified why it is necessary to add source information to
5275 GstRTSPThreadImpl. See the reported bug in GLib:
5276 https://bugzilla.gnome.org/show_bug.cgi?id=720186
5277 for more information.
5278 https://bugzilla.gnome.org/show_bug.cgi?id=761702
5280 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
5282 * examples/Makefile.am:
5283 examples: Clean up CFLAGS/LDADD even more
5284 The internal .la should come first and is part of LDADD, as is
5287 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5289 * examples/Makefile.am:
5290 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5292 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5294 * gst/rtsp-server/Makefile.am:
5295 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5297 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5299 * gst/rtsp-server/rtsp-client.c:
5300 * gst/rtsp-server/rtsp-media-factory.c:
5301 * gst/rtsp-server/rtsp-media-factory.h:
5302 * gst/rtsp-server/rtsp-media.c:
5303 * gst/rtsp-server/rtsp-media.h:
5304 * gst/rtsp-server/rtsp-sdp.c:
5305 * gst/rtsp-server/rtsp-stream.c:
5306 * gst/rtsp-server/rtsp-stream.h:
5307 rtsp-server: Implement clock signalling according to RFC7273
5308 For NTP and PTP clocks we signal the actual clock that is used and signal
5309 the direct media clock offset.
5310 For all other clocks we at least signal that it's the local sender clock.
5311 This allows receivers to know which clock was used to generate the media and
5312 its RTP timestamps. Receivers can then implement network synchronization,
5313 either absolute or at least relative by getting the sender clock rate directly
5314 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5316 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5318 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5320 * gst/rtsp-sink/gstrtspclientsink.c:
5321 rtspclientsink: Add support for setting the multicast interface
5322 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5324 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5326 * gst/rtsp-server/rtsp-media-factory.c:
5327 * gst/rtsp-server/rtsp-media-factory.h:
5328 * gst/rtsp-server/rtsp-media.c:
5329 * gst/rtsp-server/rtsp-media.h:
5330 * gst/rtsp-server/rtsp-stream.c:
5331 * gst/rtsp-server/rtsp-stream.h:
5332 rtsp-media: Add support for setting the multicast interface
5333 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5335 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5337 * gst/rtsp-sink/gstrtspclientsink.c:
5338 rtspclientsink: use new gst_element_class_add_static_pad_template()
5339 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5341 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5346 === release 1.8.0 ===
5348 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5354 * gst-rtsp-server.doap:
5357 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5359 * gst/rtsp-server/rtsp-stream.c:
5360 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5361 This would get us NO_PREROLL in the bin again and break seeking.
5362 Thanks to Carlos Rafael Giani for helping to debug this!
5363 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5365 === release 1.7.91 ===
5367 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5373 * gst-rtsp-server.doap:
5376 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5378 * gst/rtsp-server/rtsp-stream.c:
5379 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5380 Without this, RECORD pipelines are broken because
5381 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5382 added later. Previously it was there earlier and due to NO_PREROLL caused the
5383 pipeline to preroll immediately
5384 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5385 as the corresponding code previously was only for PLAY pipelines.
5386 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5388 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5390 * gst/rtsp-server/rtsp-stream.c:
5391 rtsp-stream: Fix typo in the docstring
5392 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5394 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5396 * gst/rtsp-server/rtsp-stream.c:
5397 rtsp-stream: Disable multicast loopback for all our sockets
5398 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5399 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5400 loopback setting on the socket... while udpsink does which unfortunately has
5401 no effect here on Windows but on Linux.
5402 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5404 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5406 * tests/check/gst/stream.c:
5407 stream tests: added new tests
5408 Test a case when the address pool only contains multicast addresses
5409 and the client is requesting unicast udp.
5410 Added tests for multicast ports allocation.
5411 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5413 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5415 * gst/rtsp-server/rtsp-stream.c:
5416 rtsp-stream: Only bind multicast sockets to ANY on Windows
5417 On Linux it is still needed to bind to the multicast address
5418 to filter out random other packets, while on Windows binding
5419 to multicast addresses just fails.
5421 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5423 * gst/rtsp-server/rtsp-stream.c:
5424 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5425 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5426 addresses, which is something that used to work before. For unicast addresses
5427 if the pool contains none, we just allocate them as if there is no pool at
5429 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5431 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5433 * gst/rtsp-server/rtsp-client.c:
5434 * gst/rtsp-server/rtsp-stream.c:
5435 rtsp-server: Fix indentation
5437 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5439 * gst/rtsp-server/rtsp-stream.c:
5440 rtsp-stream: Don't bind the sockets to multicast addresses
5441 This works on Linux but fails completely on Windows. You're supposed
5442 to bind to ANY and then join the multicast group.
5443 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5445 === release 1.7.90 ===
5447 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5453 * gst-rtsp-server.doap:
5456 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5459 Automatic update of common submodule
5460 From b64f03f to 6f2d209
5462 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5464 * gst/rtsp-sink/gstrtspclientsink.c:
5465 * tests/check/gst/rtspclientsink.c:
5466 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5467 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5469 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5471 * tests/check/gst/media.c:
5472 * tests/check/gst/rtspclientsink.c:
5473 * tests/check/gst/rtspserver.c:
5474 * tests/check/gst/stream.c:
5475 tests: unit test fixes
5476 Removed port allocation test from the media suite.
5477 The port allocation failure is now in the stream suite.
5479 Make sure that the media is suspended after the DESCRIBE request
5480 before reconfiguring the UDP sinks.
5482 In the RECORD case we have to set async property to false
5483 for the appsink element in the test in order to make sure
5484 that the media pipeline doesn't hang in start_preroll().
5485 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5487 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5489 * gst/rtsp-server/rtsp-client.c:
5490 * gst/rtsp-server/rtsp-stream.c:
5491 * gst/rtsp-server/rtsp-stream.h:
5492 rtsp-stream: postpone UDP socket allocation until SETUP
5493 Postpone the allocation of the UDP sockets until we know
5494 what transport has been chosen by the client.
5495 Both unicast and multicast UDP sources are created in one
5497 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5499 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5501 * gst/rtsp-server/rtsp-stream.c:
5502 rtsp-stream: postpone the creation of the UDP sources
5503 Code refactoring: allocate the UDP ports after the sender and
5504 the reciver parts have been created.
5505 We postpone the creation of the UDP sources until the UDP
5506 ports have been allocated.
5507 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5509 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5511 * gst/rtsp-server/rtsp-stream.c:
5512 rtsp-stream: added function for setting UDP sources to PLAYING state
5513 Code refactoring: Introduced a function for setting UDP sources
5515 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5517 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5519 * gst/rtsp-server/rtsp-stream.c:
5520 rtsp-stream: added function for creating and configuring UDP sources
5521 Code refactoring: create and configure UDP sources in a separate function.
5522 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5524 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5526 * gst/rtsp-server/rtsp-stream.c:
5527 rtsp-stream: added function for RTP/RTCP socket configuration
5528 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5529 in a separate function.
5530 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5532 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5534 * gst/rtsp-server/rtsp-stream.c:
5535 rtsp-stream: added function for creating and configuring UDP sinks
5536 Code refactoring: create and configure UDP sinks in a separate function.
5537 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5539 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5541 * gst/rtsp-server/rtsp-stream.c:
5542 rtsp-stream: added helper function for creating the sender/receiver parts
5543 Code refactoring: introduced helper function for creating
5544 the receiver and the sender parts of the streaming pipeline.
5545 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5547 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5552 === release 1.7.2 ===
5554 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5560 * gst-rtsp-server.doap:
5563 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5565 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5566 uninstalled.pc: add support for non libtool build systems
5567 Currently the .la path is provided which requires to use libtool as
5568 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5569 It is fine as long as the application is built using libtool.
5570 So currently it is not possible to compile a GStreamer application
5571 within gst-uninstalled with CMake or other build system different
5573 This patch allows to do the following in gst-uninstalled env:
5574 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5575 gstreamer-rtsp-server-1.0)
5576 Previously it required to prepend libtool --mode=link
5577 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5579 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5581 * gst/rtsp-sink/gstrtspclientsink.c:
5582 rtspclientsink: remove check for impossible condition
5583 Goto error label checks stream to see if it needs to be unreferenced before
5584 returning, but this goto jumps happens before the stream is ever set, so it
5585 will always be NULL in this error label.
5588 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5590 * gst/rtsp-sink/gstrtspclientsink.c:
5591 rtspclientsink: clean switch statements
5592 Coverity demands for fallthrough statements to be clearly commented,
5593 to distinguish from accidental fall throughs. And it also needs all
5594 cases to finish with a break, even if the break is never going to be
5595 executed like in the case of a continue jump.
5599 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5601 * tests/check/Makefile.am:
5602 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5603 To get the CK_DEFAULT_TIMEOUT defined for all tests
5604 Also removes a 120 seconds timeout that was set as default
5605 explicitly in this module
5606 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5608 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5612 Automatic update of common submodule
5613 From 86e4663 to b64f03f
5615 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5617 * gst/rtsp-server/rtsp-media.c:
5618 rtsp-media: fix state_lock not locked again when preroll fails
5619 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5621 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5624 configure: Move plugin specific flags below all the others
5625 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5626 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5628 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5630 * gst/rtsp-sink/gstrtspclientsink.c:
5631 rtspclientsink: Simplify slightly using new -base API
5632 Use the new Mikey and SDP API in the base plugins libs
5633 to simplify some code.
5634 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5636 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5641 * gst/rtsp-sink/Makefile.am:
5642 * gst/rtsp-sink/gstrtspclientsink.c:
5643 * gst/rtsp-sink/gstrtspclientsink.h:
5644 * gst/rtsp-sink/plugin.c:
5645 * tests/check/Makefile.am:
5646 * tests/check/gst/rtspclientsink.c:
5647 rtspsink: Add rtspclientsink element
5648 Add an rtspclientsink element that accepts streams for which
5649 there is a registered payloader and sends them to
5650 an RTSP server using RECORD.
5651 Sending is synchronised to the pipeline clock. Payload-types
5652 are automatically selected. The 'new-payloader' signal is fired
5653 for custom configuration of payloaders when they are created.
5654 Can now stream a movie like this:
5656 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5657 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5659 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5660 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5661 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5663 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5665 * gst/rtsp-server/rtsp-stream.c:
5666 * gst/rtsp-server/rtsp-stream.h:
5667 rtsp-stream: Add functions for using rtsp-stream from the client
5668 Add a boolean to indicate that the rtsp-stream is running on the
5669 'client' side of an RTSP connection, for sending streams via
5670 RECORD. In that case, the roles of the client/server ports
5671 in transport setup are swapped.
5672 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5674 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5676 * gst/rtsp-server/rtsp-sdp.c:
5677 * gst/rtsp-server/rtsp-sdp.h:
5678 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5679 A new function that adds info from a GstRTSPStream into an SDP message.
5680 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5682 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5684 * gst/rtsp-server/rtsp-media.c:
5685 rtsp-media: Fix mutex beeing unlocked while they should be locked
5686 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5688 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5690 * gst/rtsp-server/rtsp-media-factory.c:
5691 rtsp-media-factory: add missing break in "clock" property setter
5694 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5696 * gst/rtsp-server/rtsp-stream.c:
5697 rtsp-stream: fixed assert during update transport
5698 When RTSP server trying update transport during multicast, it throws an
5699 assert. The assert is thrown because it is trying to get the parent of
5700 an non-existing funnel element.
5701 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5703 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5705 * gst/rtsp-server/rtsp-permissions.h:
5706 * gst/rtsp-server/rtsp-thread-pool.h:
5707 * gst/rtsp-server/rtsp-token.h:
5708 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5709 gtk-doc can handle static inline functions just fine these days,
5710 there's no need for this stuff any more.
5712 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5714 * gst/rtsp-server/rtsp-media.c:
5715 * gst/rtsp-server/rtsp-sdp.c:
5716 sdp: replace duplicated codes to call new base sdp apis
5717 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5719 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5721 * examples/test-netclock.c:
5722 test-netclock: Use the new API to configure a clock directly
5724 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5726 * gst/rtsp-server/rtsp-media-factory.c:
5727 * gst/rtsp-server/rtsp-media-factory.h:
5728 * gst/rtsp-server/rtsp-media.c:
5729 * gst/rtsp-server/rtsp-media.h:
5730 rtsp-media: Add API to directly configure a clock on the media pipelines
5732 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5734 * gst/rtsp-server/rtsp-media.c:
5735 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5737 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5739 * gst/rtsp-server/rtsp-media-factory.c:
5740 rtsp-media-factory: Add FIXME for 2.0
5742 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5744 * gst/rtsp-server/rtsp-stream.c:
5745 rtsp-stream: Fix indentation
5747 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5749 * gst/rtsp-server/rtsp-media.c:
5750 rtsp-media: Do not prepare media after media times out
5751 Deferred calls to start_prepare() can be deferred past the point until
5752 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5753 prepared to wait. Previously there was no lock and no check for this
5754 situation. This meant that a media could be prepared and unprepared
5755 simultaneously by two different threads. Now a lock is in place and a
5756 suitable check is done.
5757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5759 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5761 * gst/rtsp-server/rtsp-client.c:
5762 * gst/rtsp-server/rtsp-media-factory.c:
5763 * gst/rtsp-server/rtsp-media-factory.h:
5764 * gst/rtsp-server/rtsp-media.c:
5765 * gst/rtsp-server/rtsp-media.h:
5766 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5767 Without TEARDOWN it might be desireable to keep the media running and continue
5768 sending data to the client, even if the RTSP connection itself is
5770 Only do this for session medias that have only UDP transports. If there's at
5771 least on TCP transport, it will stop working and cause problems when the
5772 connection is disconnected.
5773 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5775 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5780 === release 1.7.1 ===
5782 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5788 * gst-rtsp-server.doap:
5791 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5794 configure: Make -Bsymbolic check work with clang.
5795 Update the -Bsymbolic check with the version glib has. This version
5797 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5799 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5801 * gst/rtsp-server/rtsp-session-pool.c:
5802 rtsp-session-pool: Avoid dollar sign ($) in session ids
5803 Live555 in VLC strips off dollar signs and then gets very confused,
5804 we don't loose too much entropy by just skipping it.
5806 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5808 * gst/rtsp-server/rtsp-address-pool.h:
5809 * gst/rtsp-server/rtsp-auth.h:
5810 * gst/rtsp-server/rtsp-client.h:
5811 * gst/rtsp-server/rtsp-media-factory-uri.h:
5812 * gst/rtsp-server/rtsp-media-factory.h:
5813 * gst/rtsp-server/rtsp-media.h:
5814 * gst/rtsp-server/rtsp-mount-points.h:
5815 * gst/rtsp-server/rtsp-permissions.h:
5816 * gst/rtsp-server/rtsp-server.h:
5817 * gst/rtsp-server/rtsp-session-media.h:
5818 * gst/rtsp-server/rtsp-session-pool.h:
5819 * gst/rtsp-server/rtsp-session.h:
5820 * gst/rtsp-server/rtsp-stream-transport.h:
5821 * gst/rtsp-server/rtsp-stream.h:
5822 * gst/rtsp-server/rtsp-thread-pool.h:
5823 * gst/rtsp-server/rtsp-token.h:
5824 rtsp-server: Add g_autoptr() support to all types
5825 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5827 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5829 * gst/rtsp-server/rtsp-stream.c:
5830 rtsp-stream: fixed valgrind error
5831 Fixed the valgrind error in unit test. The UDP source created during
5832 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5834 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5836 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5840 Automatic update of common submodule
5841 From b319909 to 86e4663
5843 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5845 * gst/rtsp-server/rtsp-client.c:
5846 rtsp-client: suspend media during setup request
5847 SETUP request from clients needs to suspend the media to clear the
5848 prerolled buffers. Otherwise it will not affect the prerolled buffer
5849 and the prerolled buffers will be incorrect (for example block-size
5850 from setup request will not affect the prerolled buffer unless the
5851 media is suspended).
5852 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5854 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5856 * gst/rtsp-server/rtsp-stream.c:
5857 rtsp-stream: create stream pipeline based on transport
5858 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5859 only UDP is set as the transport protocol, it will not add the extra tee
5860 or queue element to the pipeline. Both these elements will be added, if
5861 it supports both TCP and UDP protocols. This improves the pipeline
5862 performance when one protocol is present.
5863 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5865 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5867 * gst/rtsp-server/rtsp-stream.c:
5868 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5869 Adding them when not needed will start some logic inside rtpbin that might be
5870 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5871 would start up a rtpjitterbuffer and behave in weird ways.
5872 We still set up the UDP sources for RTP receiving for a sender media to be
5873 able to receive any packets sent by the client for NAT traversal. They will
5874 all go to a fakesink though.
5875 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5876 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5877 receive ASYNC_DONE after a seek.
5878 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5880 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5882 * gst/rtsp-server/rtsp-stream.c:
5883 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5884 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5885 Previously we were only setting this for sender sockets, which caused looped
5886 back packets to be received on Windows if a multicast transport was used.
5888 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5890 * examples/test-record-auth.c:
5891 * examples/test-record.c:
5892 examples: Actually use the provided port in the record examples
5894 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5896 * examples/test-record-auth.c:
5897 test-record-auth: Add the option to build in TLS support
5899 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5901 * examples/test-auth.c:
5902 test-auth: Use an 'anonymous' user for unauthenticated default
5903 There's a comment on one of the resources that 'user' and 'admin'
5904 shouldn't even be able to see it, but they can if the default
5905 token is 'admin2', since that gives them access anyway.
5907 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5909 * examples/.gitignore:
5910 * examples/Makefile.am:
5911 * examples/test-record-auth.c:
5912 Add test-record-auth example
5914 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5916 * gst/rtsp-server/rtsp-client.c:
5917 * tests/check/gst/client.c:
5918 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5920 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5922 * gst/rtsp-server/rtsp-server.c:
5923 rtsp-server: Change the logic so we don't pop a NULL context
5924 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5925 will sometimes fail. This call is made before any context is pushed
5926 resulting in an attempt to pop a NULL context.
5927 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5929 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5931 * tests/check/gst/rtspserver.c:
5932 rtspserver: Add udp-mcast transport SETUP test
5933 Refactor utility functions in the test file so they can handle
5934 more than UDP and TCP as lower transport.
5935 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5937 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5939 * gst/rtsp-server/rtsp-stream.c:
5940 rtsp-stream: Always unref return value of gst_object_get_parent()
5941 Fixes a leak of a GstBin in the udp-mcast case.
5942 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5944 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5947 Automatic update of common submodule
5948 From b99800a to b319909
5950 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5953 Use new GST_ENABLE_EXTRA_CHECKS #define
5954 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5956 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5959 Automatic update of common submodule
5960 From 6babecd to b99800a
5962 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5965 Update GLib dependency to 2.40.0
5967 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5969 * examples/test-mp4.c:
5970 * gst/rtsp-server/rtsp-stream.c:
5971 stream: listen to sender ssrc signals
5972 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5974 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5977 common: update for new suppression
5978 Makes check-valgrind pass with glib 2.46
5980 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5982 * gst/rtsp-server/rtsp-media.c:
5983 rtsp-media: Take reference to media that will be prepared
5984 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5985 Later on a g_idle_source_new() is created and a pointer to the media
5986 object is passed as user data. If the media is freed before the idle
5987 source is dispatched the media object pointer is invalid, but the idle
5988 source callback expects it to still be valid. To fix this a reference to
5989 the media object is taken when registering the source callback function
5990 and a corresponding release of the reference is done when the souce is
5992 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5994 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5996 * examples/test-launch.c:
5997 * examples/test-mp4.c:
5998 * examples/test-ogg.c:
5999 * examples/test-record.c:
6000 * examples/test-uri.c:
6001 rtsp-server: Fix memory leaks when context parse fails
6002 When g_option_context_parse fails, context and error variables are not getting free'd
6003 which results in memory leaks. Free'ing the same.
6004 And replacing g_error_free with g_clear_error, which checks if the error being passed
6005 is not NULL and sets the variable to NULL on free'ing.
6006 https://bugzilla.gnome.org/show_bug.cgi?id=753863
6008 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
6013 === release 1.6.0 ===
6015 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6021 * gst-rtsp-server.doap:
6024 === release 1.5.91 ===
6026 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
6032 * gst-rtsp-server.doap:
6035 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
6037 * docs/libs/gst-rtsp-server-sections.txt:
6038 * gst/rtsp-server/rtsp-stream.c:
6039 stream: fix docs for recently-added get/set_buffer_size API
6040 https://bugzilla.gnome.org/show_bug.cgi?id=749095
6042 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
6044 * gst/rtsp-server/rtsp-media.c:
6045 rtsp-media: Don't crash on encrypted RTX SDP
6046 In parse_keymgmt(), don't mutate the input string that's been passed
6047 as const, especially since we might need the original value again if
6048 the same key info applies to multiple streams (RTX, for example).
6049 https://bugzilla.gnome.org/show_bug.cgi?id=754753
6051 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
6053 * examples/test-mp4.c:
6054 test-mp4: Support filenames with spaces in them. Error out on too few arguments
6056 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
6058 * examples/test-record.c:
6059 test-record: Check parameter count and print out help
6060 If no launch pipeline was supplied, print out some help
6062 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
6064 * gst/rtsp-server/rtsp-media.c:
6065 * gst/rtsp-server/rtsp-stream.c:
6066 * gst/rtsp-server/rtsp-stream.h:
6067 rtsp-stream: Implement UDP buffer size setting.
6068 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
6070 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
6071 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
6073 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
6075 * gst/rtsp-server/rtsp-media.h:
6076 rtsp-media: Fix small typo causing gtk-doc to complain
6078 === release 1.5.90 ===
6080 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
6086 * gst-rtsp-server.doap:
6089 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6091 * gst/rtsp-server/rtsp-media-factory.c:
6092 media-factory: get port number through gst_rtsp_url_get_port
6093 https://bugzilla.gnome.org/show_bug.cgi?id=753473
6095 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
6097 * tests/check/gst/media.c:
6098 media-test: Removing unnecessary assertion
6099 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6101 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6103 * gst/rtsp-server/rtsp-server.c:
6104 Document that source keeps a ref on server until it's destroyed
6105 https://bugzilla.gnome.org/show_bug.cgi?id=749227
6107 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6109 * tests/check/gst/media.c:
6110 media-test: Test for multiple dynamic payload
6111 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6113 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6115 * gst/rtsp-server/rtsp-media.c:
6116 media: Only add fakesink once per pipeline
6117 The intention is to prevent going PLAYING state before pads are created.
6118 If there was mutilple dynamic payload, it would leak few fakesink and
6119 actually prevent from ever reaching playing state.
6120 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6122 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6124 * gst/rtsp-server/rtsp-media.c:
6125 Revert "rtsp-media: Only add 1 fakesink per pipeline"
6126 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
6128 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6130 * gst/rtsp-server/rtsp-media.c:
6131 rtsp-media: Only add 1 fakesink per pipeline
6132 There should be only one fakesink per pipeline, not per dynpay. This
6133 would lead to element naming clash.
6135 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
6137 * gst/rtsp-server/rtsp-media.c:
6138 rtsp-media: assertion error due to wrong condition check
6139 In media to caps function, reserved_keys array is being used for variable i,
6140 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
6141 changed it to variable j
6142 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6144 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
6146 * gst/rtsp-server/rtsp-media.c:
6147 rtsp-media: Strip keys from the fmtp that we use internally in our caps
6148 Skip keys from the fmtp, which we already use ourselves for the
6149 caps. Some software is adding random things like clock-rate into
6150 the fmtp, and we would otherwise here set a string-typed clock-rate
6151 in the caps... and thus fail to create valid RTP caps
6152 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6154 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6156 * gst/rtsp-server/rtsp-thread-pool.c:
6157 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
6158 https://bugzilla.gnome.org/show_bug.cgi?id=752640
6160 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
6163 Automatic update of common submodule
6164 From f74b2df to 9aed1d7
6166 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
6171 === release 1.5.2 ===
6173 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
6179 * gst-rtsp-server.doap:
6182 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
6184 * gst/rtsp-server/rtsp-client.c:
6185 * gst/rtsp-server/rtsp-client.h:
6186 * tests/check/gst/client.c:
6187 rtsp-client: allow application to decide what requirements are supported
6188 Add "check-requirements" signal and vfunc to allow application
6189 (and subclasses) to check the requirements.
6190 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
6191 https://bugzilla.gnome.org/show_bug.cgi?id=749417
6193 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6196 Automatic update of common submodule
6197 From 6015d26 to f74b2df
6199 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6201 * gst/rtsp-server/rtsp-media.c:
6202 rtsp-media: Always use real payloader when creating streams
6203 A bin that contains the real payloader might be used as payloader. In this
6204 case we have to get the real payloader for the various properties it provides.
6205 Example use cases for this are bins that payload some media and then have
6206 additional elements that add metadata or RTP extension headers to the stream.
6207 https://bugzilla.gnome.org/show_bug.cgi?id=750800
6209 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
6211 * examples/test-netclock-client.c:
6212 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
6214 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
6216 * examples/test-netclock-client.c:
6217 * examples/test-netclock.c:
6218 test-netclock: Use new ntp-time-source property on rtpbin
6219 Select the clock time to be used as NTP time source. This allows proper
6220 synchronization between receivers, independent of sharing base times, and just
6221 requires them to use the same clock.
6223 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6225 * examples/test-netclock-client.c:
6226 * examples/test-netclock.c:
6227 test-netclock: Setting the same base time on sender and receiver is not necessary
6228 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
6230 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6232 * gst/rtsp-server/rtsp-stream.c:
6233 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
6234 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6236 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6238 * docs/libs/gst-rtsp-server.types:
6239 docs: add missing types
6240 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6242 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6244 * docs/libs/gst-rtsp-server-sections.txt:
6245 docs: add missing apis
6246 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6248 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
6250 * examples/test-netclock-client.c:
6251 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
6253 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6255 * docs/libs/gst-rtsp-server-sections.txt:
6256 * gst/rtsp-server/rtsp-auth.c:
6257 * gst/rtsp-server/rtsp-auth.h:
6258 GstRTSPAuth: Add client certificate authentication support
6259 https://bugzilla.gnome.org/show_bug.cgi?id=750471
6261 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
6263 * examples/test-netclock-client.c:
6264 test-netclock-client: Use new GstClock API to wait for clock synchronization
6266 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
6268 * examples/test-netclock-client.c:
6269 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
6270 A mainloop is needed to get glimagesink to display something on OSX, and
6271 the source-setup signal just makes things a little bit easier.
6273 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
6276 Automatic update of common submodule
6277 From d9a3353 to 6015d26
6279 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
6282 Automatic update of common submodule
6283 From d37af32 to d9a3353
6285 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6288 Automatic update of common submodule
6289 From 21ba2e5 to d37af32
6291 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6294 Automatic update of common submodule
6295 From c408583 to 21ba2e5
6297 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6299 * docs/libs/Makefile.am:
6300 docs: remove variables that we define in the snippet from common
6301 This is syncing our Makefile.am with upstream gtkdoc.
6303 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6306 Automatic update of common submodule
6307 From 44a3517 to c408583
6309 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6314 === release 1.5.1 ===
6316 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6322 * gst-rtsp-server.doap:
6325 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6327 * gst/rtsp-server/rtsp-client.c:
6328 rtsp-client: No flush during Teardown.
6329 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6330 backlog is empty it can happen that just a part of a message will be
6331 sent and rest is in backlog queue. If then flush during teardown
6332 just a part of message will be sent.This can lead to client miss
6333 teardown response since it expect to get the last part of message.
6334 The flushing during teardown was introduced to fix a deadlock that now
6335 is fixed more generally in handle_request by temporary setting backlog
6337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6339 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6341 * tests/check/Makefile.am:
6342 tests: Use AM_TESTS_ENVIRONMENT
6343 Needed by the new automake test runner and the
6344 current version of the common submodule.
6346 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6348 * gst/rtsp-server/rtsp-media.h:
6349 * gst/rtsp-server/rtsp-stream.h:
6350 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6352 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6354 * gst/rtsp-server/rtsp-media.c:
6355 rtsp-media: Mark some more functions static
6357 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6359 * gst/rtsp-server/rtsp-media.c:
6360 rtsp-media: Only unblock the media in suspend() when actually changing the state
6361 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6363 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6365 * examples/test-video-rtx.c:
6366 examples: Use AVPF profile for the RTX example
6368 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6370 * gst/rtsp-server/rtsp-sdp.c:
6371 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6373 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6375 * gst/rtsp-server/rtsp-stream.c:
6376 rtsp-stream: get valid clock-rate from last-sample
6377 clock-rate in last-sample's caps is integer, not unsigned.
6378 To get this value properly, variable needs to be type-casted to int.
6379 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6381 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6385 autogen.sh: only run autopoint if gettext requested in configure.ac
6386 Not just because there happens to be a po directory.
6387 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6389 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6392 Revert "configure.ac: uncomment gettext version setup"
6393 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6394 We don't need a gettext setup here and there's no po
6395 directory either, so no reason why autopoint would be
6396 run in the first place.
6397 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6399 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6401 * examples/test-multicast.c:
6402 * examples/test-multicast2.c:
6403 * examples/test-sdp.c:
6404 * examples/test-video-rtx.c:
6405 * examples/test-video.c:
6406 * tests/test-cleanup.c:
6407 * tests/test-reuse.c:
6408 Fix timeout function signatures across tests and examples
6410 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6412 * tests/check/Makefile.am:
6413 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6414 Make sure the test environment is set up.
6415 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6417 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6420 configure: bump automake requirement to 1.14 and autoconf to 2.69
6421 This is only required for builds from git, people can still
6422 build tarballs if they only have older autotools.
6423 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6425 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6428 configure.ac: uncomment gettext version setup
6429 Fixes autogen.sh. It would run autopoint, which would complain
6430 that it could not find the gettext version in configure.ac.
6431 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6433 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6435 * examples/test-video-rtx.c:
6436 test-video-rtx: set exact payload type to PCMA payloader
6437 Setting wrong payload type causes failure to do retransmission through audio stream
6438 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6440 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6442 * gst/rtsp-server/rtsp-media.c:
6443 * gst/rtsp-server/rtsp-stream.c:
6444 * gst/rtsp-server/rtsp-stream.h:
6445 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6446 Because of duplicated g_signal_connect for request-aux-sender signal,
6447 wrong stream pointer is passed to the signal handler.
6448 Instead of passing each stream, pass stream array and get the relevant stream.
6449 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6451 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6455 Update autogen.sh to latest version from common
6456 Fixes build after aclocal_check etc. helpers have been removed.
6458 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6461 Automatic update of common submodule
6462 From bc76a8b to c8fb372
6464 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6466 * gst/rtsp-server/rtsp-stream.c:
6467 rtsp-stream: Limit the queues to 1 buffer
6468 We only need them to be able to pre-roll, queueing up more data here
6469 is only going to harm latency and memory usage.
6471 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6473 * gst/rtsp-server/rtsp-stream.c:
6474 rtsp-stream: Update comment and ASCII art to the latest code
6475 We have a queue in front of the udpsink too to prevent the pipeline from
6478 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6480 * gst/rtsp-server/rtsp-stream.c:
6481 rtsp-media: Properly return first rtptime
6482 Instead we where returning first GstBuffer timestamp. This would result
6483 in clock skew and unwanted behaviour in RTSP playback.
6484 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6486 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6488 * gst/rtsp-server/rtsp-stream.c:
6489 rtsp-stream: Don't leave buffer mapped
6490 If the seq is NULL, the RTP buffer was left mapped. We should always
6493 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6498 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6500 * gst/rtsp-server/rtsp-media-factory.c:
6501 * tests/check/gst/client.c:
6502 Fix double semicolons
6504 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6506 * gst/rtsp-server/rtsp-stream.c:
6507 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6508 This gives more accurate values than asking the payloader. There might be
6509 queueing happening between the payloader and the sink.
6510 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6512 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6514 * gst/rtsp-server/rtsp-media.c:
6515 rtsp-media: Don't seek for PLAY if the position will not change
6516 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6518 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6520 * gst/rtsp-server/rtsp-media.c:
6521 rtsp-media: Don't include payload type in the caps for framesize
6522 When the sdp media attribute framesize are converted to caps
6523 the <payload> should not be included.
6524 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6525 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6527 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6529 * gst/rtsp-server/rtsp-sdp.c:
6530 rtsp-sdp: add payload type to the sdp framesize attribute
6531 The sdp framesize attribute is desribed in RFC6064. It is specified
6532 for payloading of H263 and has the following form
6533 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6534 should be added to the caps in a payloader and the <payload type> should
6535 be added by the rtsp-server.
6536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6538 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6540 * examples/test-uri.c:
6541 examples: test-uri: fix tainted variable
6542 Insignificant but this keeps Coverity happy.
6545 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6547 * examples/.gitignore:
6548 * examples/Makefile.am:
6549 * examples/test-netclock-client.c:
6550 * examples/test-netclock.c:
6551 examples: Add a simple example of network synch for live streams.
6552 An example server and client that works for synchronising live streams
6553 only - as it can't support pause/play.
6555 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6557 * gst/rtsp-server/rtsp-media-factory.c:
6558 * gst/rtsp-server/rtsp-media-factory.h:
6559 rtsp-media-factory: Add functions to set/get the media gtype
6560 Allow specifying the GType of a GstRtspMedia subclass to create
6561 as a simpler way to get the factory to create a custom
6562 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6564 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6566 * gst/rtsp-server/rtsp-media.c:
6567 rtsp-media: fix double unlock in _get_buffer_size()
6568 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6569 because of double g_mutex_unlock () usage.
6570 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6572 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6574 * gst/rtsp-server/rtsp-session-pool.c:
6575 * gst/rtsp-server/rtsp-session.c:
6576 * gst/rtsp-server/rtsp-session.h:
6577 rtsp-session: Use monotonic time for RTSP session timeout
6578 Changed RTSP session timeout handling to monotonic time
6579 and deprecating the API for current system time.
6580 This fixes timeouts when the system time changes.
6581 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6583 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6585 * gst/rtsp-server/rtsp-client.c:
6586 * gst/rtsp-server/rtsp-media.c:
6587 rtsp-client: Only error out in PLAY if seeking actually failed
6588 If the media was just not seekable, we continue from whatever position we are
6589 and let the client decide if that is what is wanted or not.
6590 Only if the actual seek failed, we can't really recover and should error out.
6592 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6594 * gst/rtsp-server/rtsp-stream.c:
6595 rtsp-stream: Add necessary queues between tee and multiudpsink
6596 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6598 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6600 * gst/rtsp-server/rtsp-client.c:
6601 * gst/rtsp-server/rtsp-media.c:
6602 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6603 Instead error out properly the same way as if the SEEKING query already
6606 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6608 * gst/rtsp-server/rtsp-stream.h:
6609 rtsp-stream: minor code formatting fix
6611 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6613 * gst/rtsp-server/rtsp-media.c:
6614 rtsp-media: fix logic for collect_streams
6615 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6616 all streams it knows if it got any, and can check if the transport mode is OK.
6619 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6621 * gst/rtsp-server/rtsp-media.c:
6622 rtsp-media: Don't set the transport mode based on what elements we find
6623 Just print a warning if the one that was set before disagrees with what
6624 elements we found. It must already be set to something before as this
6625 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6626 and we would reject ANNOUNCE if the RECORD flag was not set.
6628 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6630 * tests/check/gst/rtspserver.c:
6631 tests: rtspserver: rename shadowed variable
6632 We have two different 'sink' variables here,
6633 rename one of them for clarity.
6635 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6637 * gst/rtsp-server/rtsp-client.c:
6638 rtsp-client: fix awkward if clause
6640 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6642 * examples/test-uri.c:
6643 examples: test-uri: improve uri argument handling and accept file names
6644 Print an error if the argument passed is not a URI and can't
6645 be converted into one, or no arguments have been provided.
6647 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6649 * examples/test-uri.c:
6650 examples: test-uri: don't remove mount point after 10 seconds
6651 It's very irritating when trying to test stuff repeatedly
6652 and serves no real purpose other than showing that it can
6655 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6657 * examples/.gitignore:
6658 examples: add new test-record to .gitignore
6660 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6662 * examples/test-record.c:
6663 * gst/rtsp-server/rtsp-client.c:
6664 * gst/rtsp-server/rtsp-media-factory.c:
6665 * gst/rtsp-server/rtsp-media-factory.h:
6666 * gst/rtsp-server/rtsp-media.c:
6667 * gst/rtsp-server/rtsp-media.h:
6668 * tests/check/gst/rtspserver.c:
6669 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6671 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6673 * examples/test-record.c:
6674 test-record: Set latency for playback-style example to 2s instead of 200ms
6676 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6678 * tests/check/gst/rtspserver.c:
6679 tests: add some unit tests for ANNOUNCE and RECORD
6680 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6682 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6684 * gst/rtsp-server/rtsp-client.c:
6685 rtsp-client: fix a couple of leaks in handle_announce
6687 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6689 * gst/rtsp-server/rtsp-media-factory.c:
6690 * gst/rtsp-server/rtsp-media-factory.h:
6691 * gst/rtsp-server/rtsp-media.c:
6692 * gst/rtsp-server/rtsp-media.h:
6693 rtsp-media: Expose latency setting for setting the rtpbin latency
6695 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6697 * examples/test-record.c:
6698 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6700 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6702 * gst/rtsp-server/rtsp-stream.c:
6703 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6705 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6707 * examples/Makefile.am:
6708 * examples/test-record.c:
6709 * gst/rtsp-server/rtsp-client.c:
6710 * gst/rtsp-server/rtsp-client.h:
6711 * gst/rtsp-server/rtsp-media-factory.c:
6712 * gst/rtsp-server/rtsp-media-factory.h:
6713 * gst/rtsp-server/rtsp-media.c:
6714 * gst/rtsp-server/rtsp-media.h:
6715 * gst/rtsp-server/rtsp-session-media.c:
6716 * gst/rtsp-server/rtsp-stream.c:
6717 * gst/rtsp-server/rtsp-stream.h:
6718 Add initial support for RECORD
6719 We currently only support media that is RECORD or PLAY only, not both at once.
6720 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6722 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6724 * gst/rtsp-server/rtsp-stream.c:
6725 rtsp-stream: RTCP and RTP transport cache cookies seperated
6726 RTCP packets were not sent because the same tr_cache_cookie was used for
6727 both RTP and RTCP. So only one of the tr_cache lists were populated
6728 depending on which one was sent first. If the tr_cache list is not
6729 populated then no packets can be sent. Most often this happened to be
6730 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6731 resulted in both the tr_cache_lists to be populated regardless of which
6733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6735 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6737 * gst/rtsp-server/rtsp-stream.c:
6738 rtsp-stream: fix false compiler warning
6739 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6741 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6743 * gst/rtsp-server/rtsp-client.c:
6744 rtsp-client: log interleaved data received
6746 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6748 * gst/rtsp-server/rtsp-client.c:
6749 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6751 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6753 * gst/rtsp-server/rtsp-client.c:
6754 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6756 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6758 * gst/rtsp-server/rtsp-client.c:
6759 rtsp-client: Use a random session ID in the SDP
6760 RFC4566 Section 5.2 says that it should make the username, session id,
6761 nettype, addrtype and unicast address tuple globally unique. Always using
6762 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6763 Instead let's create a 64 bit random number, which at least brings us
6764 closer to the goal of global uniqueness.
6765 https://tools.ietf.org/html/rfc4566#section-5.2
6767 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6769 * examples/test-launch.c:
6770 * examples/test-mp4.c:
6771 * examples/test-ogg.c:
6772 * examples/test-uri.c:
6773 examples: Don't call gst_init() and gst_get_option_group()
6774 The latter calls the former at the appropriate time.
6776 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6778 * gst/rtsp-server/rtsp-client.c:
6779 rtsp-client: Drop trailing \0 of RTSP DATA messages
6780 We add a trailing \0 in GstRTSPConnection to make parsing of
6781 string message bodies easier (e.g. the SDP from DESCRIBE) but
6782 for actual data this means we have to drop it or otherwise
6783 create invalid data.
6785 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6787 * gst/rtsp-server/rtsp-stream.c:
6788 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6789 Fixes crash when two threads access handle_new_sample() at the same
6790 time, one for RTP, one for RTCP.
6791 Otherwise, when iterating over the transports cache, it might be modified by
6792 another thread at the same time if the transports cookie has changed.
6793 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6795 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6797 * gst/rtsp-server/rtsp-stream.c:
6798 rtsp-stream: Set format=TIME on our app sources for TCP
6800 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6802 * gst/rtsp-server/rtsp-session-pool.c:
6803 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6804 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6805 RFC 2326 states that session IDs may consist of alphanumeric as well as
6806 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6807 Previously the session ID was URI-escaped, this meant that any character
6808 which was not alphanumeric or any of the characters +-._~ would be
6809 percent encoded. While the RFC (surprisingly) mentions that linear white
6810 space in session IDs should be URI-escaped, it does not say anything
6811 about other characters. Moreover no white space is allowed in the
6812 session ID. Finally the percent character which is the result of
6813 URI-escaping is not allowed in a session ID.
6814 So there is no reason to do any URI-escaping, and now it is removed.
6815 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6817 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6820 Automatic update of common submodule
6821 From f2c6b95 to bc76a8b
6823 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6826 Fix 'make check' from top-level directory
6828 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6830 * examples/test-launch.c:
6831 * examples/test-mp4.c:
6832 * examples/test-ogg.c:
6833 * examples/test-uri.c:
6834 examples: Add command-line parsing and take a 'port' argument
6835 This allows users to run multiple servers on different ports for testing.
6836 Only done for examples that actually take arguments and hence are capable of
6837 outputting different streams for each instance on each port.
6838 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6840 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6842 * gst/rtsp-server/rtsp-client.c:
6843 * gst/rtsp-server/rtsp-client.h:
6844 rtsp-client: Add a send_message default signal handler
6845 This allows subclasses to easily hook into the response sending
6846 mechanism without doing everything from a signal, which seems
6847 awkward from subclasses.
6849 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6852 Automatic update of common submodule
6853 From ef1ffdc to f2c6b95
6855 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6859 configure: add --disable-examples switch
6860 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6862 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6864 * examples/.gitignore:
6865 * examples/Makefile.am:
6866 * examples/test-video-rtx.c:
6867 examples: add a retransmisison example implementing RFC4588
6868 Currently only SSRC-multiplexed rtx streams are supported
6870 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6872 * gst/rtsp-server/rtsp-stream.c:
6873 rtsp-stream: Fix some minor memory leaks
6875 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6877 * gst/rtsp-server/rtsp-media.c:
6878 rtsp-media: Some minor cleanup
6880 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6882 * gst/rtsp-server/rtsp-stream.c:
6883 rtsp-stream: Fix compiler warnings
6884 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6885 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6887 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6888 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6891 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6893 * docs/libs/gst-rtsp-server-sections.txt:
6894 * gst/rtsp-server/rtsp-media-factory.c:
6895 * gst/rtsp-server/rtsp-media-factory.h:
6896 * gst/rtsp-server/rtsp-media.c:
6897 * gst/rtsp-server/rtsp-media.h:
6898 * gst/rtsp-server/rtsp-sdp.c:
6899 * gst/rtsp-server/rtsp-stream.c:
6900 * gst/rtsp-server/rtsp-stream.h:
6901 media: implement ssrc-multiplexed retransmission support
6902 based off RFC 4588 and the server-rtpaux example in -good
6904 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6906 * gst/rtsp-server/rtsp-client.c:
6907 * gst/rtsp-server/rtsp-stream-transport.c:
6908 * gst/rtsp-server/rtsp-stream.c:
6909 rtsp: Ref transports in hash table.
6910 Also ref streams for transports.
6911 This solves a crash when reciving a rtcp after teardown but before
6913 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6915 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6918 Automatic update of common submodule
6919 From 7bb2bce to ef1ffdc
6921 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6923 * gst/rtsp-server/rtsp-client.c:
6924 client: refactor cleanup of cached media
6926 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6928 * tests/check/gst/client.c:
6930 The session leak is now fixed, lets remove those FIXME comments.
6932 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6934 * tests/check/gst/rtspserver.c:
6935 tests: Test to setup two sessions on one connection
6936 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6938 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6940 * tests/check/gst/rtspserver.c:
6941 tests: Test setup with tcp transport
6942 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6944 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6946 * gst/rtsp-server/rtsp-client.c:
6947 client: Configure transport after creating session media
6948 The default implementation of configure_client_transport() in
6949 rtsp-client uses the session media when it chooses channels for
6950 interleaved traffic.
6951 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6953 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6955 * gst/rtsp-server/rtsp-client.c:
6956 * gst/rtsp-server/rtsp-session-media.c:
6957 client: Stop caching media in client when doing setup
6958 If the media has been managed by a session media, it should not be
6959 cached in the client any longer. The GstRTSPSessionMedia object is now
6960 responsible for unpreparing the GstRTSPMedia object using
6961 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6963 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6965 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6967 * gst/rtsp-server/rtsp-stream.c:
6968 rtsp-stream: unref srtp decoder when leaving bin
6969 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6971 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6973 * gst/rtsp-server/rtsp-client.c:
6974 rtsp-client: mikey memory leaks
6975 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6977 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6980 Automatic update of common submodule
6981 From 84d06cd to 7bb2bce
6983 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6986 Parallelise 'make check-valgrind'
6988 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6991 Automatic update of common submodule
6992 From a8c8939 to 84d06cd
6994 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6997 Automatic update of common submodule
6998 From 36388a1 to a8c8939
7000 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7002 * gst/rtsp-server/rtsp-media.c:
7003 rtsp-media: deactivate media when shutting down from paused
7004 This was only done when going directly from playing.
7005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
7007 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7009 * gst/rtsp-server/rtsp-client.c:
7010 * gst/rtsp-server/rtsp-context.h:
7011 rtsp-client: add stream transport to context
7012 We add the stream transport to the context so we can get the configured
7013 client stream transport in the setup request signal.
7014 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
7016 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7018 * gst/rtsp-server/rtsp-stream.c:
7019 stream: release lock even not all transports have been removed
7020 We don't want to keep the lock even we return FALSE because not all the
7021 transports have been removed. This could lead into a deadlock.
7022 https://bugzilla.gnome.org/show_bug.cgi?id=737797
7024 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
7026 * gst/rtsp-server/rtsp-sdp.c:
7027 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
7028 These were renamed in GstRTPBasePayload in 1.0
7030 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7032 * gst/rtsp-server/rtsp-client.c:
7033 client: set session media to NULL without the lock
7034 We need to set session medias to NULL without the client lock otherwise
7035 we can end up in a deadlock if another thread is waiting for the lock
7036 and media unprepare is also waiting for that thread to end.
7037 https://bugzilla.gnome.org/show_bug.cgi?id=737690
7039 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
7041 * gst/rtsp-server/rtsp-media.c:
7042 rtsp-media: Set state to UNPREPARING in all cases
7044 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
7046 * gst/rtsp-server/rtsp-media.c:
7047 media: set state to unpreparing when unprepare is initiated
7048 https://bugzilla.gnome.org/show_bug.cgi?id=737675
7050 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
7052 * gst/rtsp-server/rtsp-client.c:
7053 rtsp-client: Remove backlog limit while processings requests
7054 If the backlog limit is kept two cases of deadlocks may be
7055 encountered when streaming over TCP. Without the backlog
7056 limit this deadlocks can not happen, at the expence of
7058 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
7060 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
7062 * gst/rtsp-server/rtsp-client.c:
7063 rtsp-client: do not free main context before rtsp watch
7064 https://bugzilla.gnome.org/show_bug.cgi?id=737110
7066 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
7068 * tests/check/gst/rtspserver.c:
7069 tests: Extend unit test timeout to accomodate for valgrind
7070 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7072 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
7074 * gst/rtsp-server/rtsp-client.c:
7075 * gst/rtsp-server/rtsp-session.c:
7076 * gst/rtsp-server/rtsp-stream-transport.c:
7077 rtsp-*: Treat sending packets to clients as keepalive
7078 As long as gst-rtsp-server can successfully send RTP/RTCP data to
7079 clients then the client must be reading. This change makes the server
7080 timeout the connection if the client stops reading.
7081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7083 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
7085 * gst/rtsp-server/rtsp-client.c:
7086 rtsp-client: Allow backlog to grow while expiring session
7087 Allow the send backlog in the RTSP watch to grow to unlimited size while
7088 attempting to bring the media pipeline to NULL due to a session
7089 expiring. Without this change the appsink element cannot change state
7090 because it is blocked while rendering data in the new_sample callback.
7091 This callback will block until it has successfully put the data into the
7092 send backlog. There is a chance that the send backlog is full at this
7093 point which means that the callback may block for a long time, possibly
7094 forever. Therefore the media pipeline may also be prevented from
7095 changing state for a long time.
7096 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7098 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
7100 * gst/rtsp-server/rtsp-client.c:
7101 rtsp-client: Make old compilers happy
7102 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
7103 Just in case that guint8 doesn't fit in a pointer. Just in case ...
7105 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
7107 * gst/rtsp-server/rtsp-client.c:
7108 client: raise the backlog limits before pausing
7109 We need to raise the backlog limits before pausing the pipeline or else
7110 the appsink might be blocking in the render method in wait_backlog() and
7111 we would deadlock waiting for paused.
7112 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
7114 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
7116 * gst/rtsp-server/rtsp-client.c:
7117 client: make define for the WATCH_BACKLOG
7118 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
7120 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
7122 * gst/rtsp-server/rtsp-client.c:
7123 client: simplify session transport handling
7124 link/unlink of the transport in a session was done to keep track of all
7125 TCP transports and to send RTP/RTCP data to the streams. We can simplify
7126 that by putting all the TCP transports in a hashtable indexed with the
7128 We also don't need to link/unlink the transports when we pause/resume
7129 the streams. The same effect is already achieved when we pause/play the
7130 media. Indeed, when we pause the media, the transport is removed from
7131 the media and the callbacks will not be called anymore.
7132 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
7134 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
7136 * gst/rtsp-server/rtsp-stream-transport.c:
7137 * gst/rtsp-server/rtsp-stream-transport.h:
7138 stream-transport: make method to handle received data
7139 Make a method to handle the data received on a channel. It sends the
7140 data to the stream of the transport on the RTP or RTCP pads based on
7143 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
7145 * examples/test-mp4.c:
7146 test: add example of dumping RTCP reports
7148 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
7150 * gst/rtsp-server/rtsp-media.c:
7151 * gst/rtsp-server/rtsp-stream.c:
7152 * gst/rtsp-server/rtsp-stream.h:
7153 rtsp-media: Make sure that sequence numbers are monotonic after pause
7154 The sequence number is not monotonic for RTP packets after pause. The
7155 reason is basepayloader generates a randon sequence number when the
7156 pipeline goes from ready to pause. With this fix generation of sequence
7157 number will be monotonic when going from pause to play request.
7158 https://bugzilla.gnome.org/show_bug.cgi?id=736017
7160 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
7162 * gst/rtsp-server/rtsp-client.c:
7163 rtsp-client: Protect saved clients watch with a mutex
7164 Fixes a crash when close() is called while merging clients
7165 in handle_tunnel(). In that case close() would destroy the
7166 watch while it is still being used in handle_tunnel().
7167 https://bugzilla.gnome.org/show_bug.cgi?id=735570
7169 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
7171 * gst/rtsp-server/rtsp-stream.c:
7172 rtsp-stream: Remove the multicast group udp sources when removing from the bin
7174 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
7176 * gst/rtsp-server/rtsp-media.c:
7177 * gst/rtsp-server/rtsp-stream.c:
7178 * gst/rtsp-server/rtsp-stream.h:
7179 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
7180 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
7181 seeking and will always continue counting the time. This leads to
7182 the NPT after a backwards seek to be something completely different
7183 to the actual seek position.
7184 https://bugzilla.gnome.org/show_bug.cgi?id=732644
7186 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
7188 * examples/test-appsrc.c:
7189 examples: fix another reference leak
7190 gst_rtsp_media_get_element() returns a new ref.
7192 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
7194 * examples/test-appsrc.c:
7195 examples: unref element after usage
7196 gst_bin_get_by_name_recurse_up() returns an element
7197 reference that must be unreffed after usage.
7198 https://bugzilla.gnome.org/show_bug.cgi?id=734546
7200 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
7202 * gst/rtsp-server/rtsp-media.c:
7203 signals: Fix copy-pasto in target-state signal offset
7205 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
7209 Makefile: Add usage of build-checks step
7210 Allows building checks without running them
7212 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
7214 * gst/rtsp-server/rtsp-stream.c:
7215 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
7216 When a UDP multicast transport is used it is expected that the server listens
7217 for RTP and RTCP packets on the multicast group with the corresponding port.
7218 Without this we will never get RTCP packets from clients in multicast mode.
7219 https://bugzilla.gnome.org/show_bug.cgi?id=732238
7221 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
7226 === release 1.4.0 ===
7228 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7234 * gst-rtsp-server.doap:
7237 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
7239 * gst/rtsp-server/rtsp-media.h:
7240 media: correct misspelled words in description
7241 https://bugzilla.gnome.org/show_bug.cgi?id=733244
7243 === release 1.3.91 ===
7245 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
7251 * gst-rtsp-server.doap:
7254 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
7256 * docs/libs/gst-rtsp-server-sections.txt:
7259 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
7261 * gst/rtsp-server/rtsp-server.c:
7262 server: implement client REMOVE filter
7264 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
7266 * gst/rtsp-server/rtsp-client.c:
7267 * gst/rtsp-server/rtsp-client.h:
7268 client: expose _close() method
7269 Expose a previously internal close method to close the client
7272 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
7274 * gst/rtsp-server/rtsp-session-pool.c:
7275 session-pool: signal session-removed outside of the lock
7276 Release the lock before emiting the session-removed signal.
7278 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
7280 * gst/rtsp-server/rtsp-client.c:
7281 * gst/rtsp-server/rtsp-server.c:
7282 * gst/rtsp-server/rtsp-session-pool.c:
7283 * gst/rtsp-server/rtsp-session.c:
7284 * gst/rtsp-server/rtsp-stream.c:
7285 filter: Release lock in filter functions
7286 Release the object lock before calling the filter functions. We need to
7287 keep a cookie to detect when the list changed during the filter
7288 callback. We also keep a hashtable to make sure we only call the filter
7289 function once for each object in case of concurrent modification.
7290 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7292 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7294 * gst/rtsp-server/rtsp-client.c:
7295 client: check if watch is set in handle_teardown()
7296 The unit tests run without a watch
7298 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7300 * tests/check/gst/client.c:
7301 client tests: send teardown to cleanup session
7303 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7305 * tests/check/gst/rtspserver.c:
7306 server tests: send teardown to cleanup session
7308 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7310 * gst/rtsp-server/rtsp-client.c:
7311 client: keep ref to client for the session removed handler
7312 This extra ref will be dropped when all client sessions have been
7313 removed. A session is removed when a client sends teardown, closes its
7314 endpoint of the TCP connection or the sessions expires.
7315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7317 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7319 * gst/rtsp-server/rtsp-client.c:
7320 * gst/rtsp-server/rtsp-session.c:
7321 * tests/check/gst/client.c:
7322 client: manage media in session as a last step
7323 Once we manage a media in a session, we can't unmanage it anymore
7324 without destroying it. Therefore, first check everything before we
7325 manage the media, otherwise if something is wrong we have no way to
7327 If we created a new session and something went wrong, remove the session
7328 again. Fixes a leak in the unit test.
7330 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7332 * examples/test-mp4.c:
7333 * examples/test-ogg.c:
7334 examples: print 'stream ready at url' for mp4 and ogg example
7336 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7338 * gst/rtsp-server/rtsp-client.c:
7339 * gst/rtsp-server/rtsp-sdp.c:
7340 rtsp: fix for MIKEY api change
7342 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7344 * gst/rtsp-server/rtsp-client.c:
7345 client: free watch context only once
7346 The watch context is freed when the source is destroyed. Avoids
7347 a CRITICAL when we try to unref the context twice.
7349 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7351 * gst/rtsp-server/rtsp-client.c:
7354 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7356 * gst/rtsp-server/rtsp-client.c:
7357 client: protect sessions with lock
7358 Protect the list of sessions with the lock.
7359 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7361 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7363 * gst/rtsp-server/rtsp-client.c:
7364 Client: keep a ref to the session
7365 Don't just keep a weak ref to the session objects but use a hard ref. We
7366 will be notified when a session is removed from the pool (expired) with
7367 the new session-removed signal.
7368 Don't automatically close the RTSP connection when all the sessions of
7369 a client are removed, a client can continue to operate and it can create
7370 a new session if it wants. If you want to remove the client from the
7371 server, you have to use gst_rtsp_server_client_filter() now.
7372 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7373 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7375 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7377 * gst/rtsp-server/rtsp-session-pool.c:
7378 * gst/rtsp-server/rtsp-session-pool.h:
7379 session-pool: add session-removed signal
7380 Add a signal to be notified when a session is removed from the pool.
7382 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7384 * gst/rtsp-server/Makefile.am:
7385 * gst/rtsp-server/rtsp-server.h:
7386 Make rtsp-server.h a single-include header, use it for G-I
7387 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7389 === release 1.3.90 ===
7391 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7397 * gst-rtsp-server.doap:
7400 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7402 * gst/rtsp-server/rtsp-stream.c:
7403 stream: crypto can be NULL
7405 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7407 * gst/rtsp-server/rtsp-client.c:
7408 * gst/rtsp-server/rtsp-media.c:
7409 * gst/rtsp-server/rtsp-mount-points.c:
7410 introspection: add missing allow-none annotations
7411 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7413 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7415 * gst/rtsp-server/rtsp-address-pool.c:
7416 * gst/rtsp-server/rtsp-media.c:
7417 * gst/rtsp-server/rtsp-session-media.c:
7418 * gst/rtsp-server/rtsp-session-pool.c:
7419 * gst/rtsp-server/rtsp-stream-transport.c:
7420 * gst/rtsp-server/rtsp-stream.c:
7421 * gst/rtsp-server/rtsp-token.c:
7422 introspection: add (nullable) annotations to return values
7423 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7425 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7427 * gst/rtsp-server/rtsp-client.c:
7428 * gst/rtsp-server/rtsp-stream.c:
7429 gi: improve annotations
7430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7432 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7434 * gst/rtsp-server/rtsp-client.c:
7435 * gst/rtsp-server/rtsp-media-factory.c:
7436 * gst/rtsp-server/rtsp-media.c:
7437 * gst/rtsp-server/rtsp-server.c:
7438 signals: use generic marshal function
7439 Use the generic C marshal function.
7440 Use more explicit type instead of G_TYPE_POINTER
7442 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7444 * gst/rtsp-server/rtsp-context.h:
7445 context: add type macro
7447 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7449 * gst/rtsp-server/rtsp-client.c:
7450 * gst/rtsp-server/rtsp-sdp.c:
7451 * gst/rtsp-server/rtsp-sdp.h:
7452 sdp: hide key length defines
7453 They don't have a namespace.
7455 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7460 === release 1.3.3 ===
7462 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7468 * gst-rtsp-server.doap:
7471 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7473 * gst/rtsp-server/rtsp-client.c:
7474 * gst/rtsp-server/rtsp-sdp.c:
7475 * gst/rtsp-server/rtsp-sdp.h:
7476 mikey: add different key length parameters
7477 Add encryption and authentication key length parameters to MIKEY. For
7478 the encoders, the key lengths are obtained from the cipher and auth
7479 algorithms set in the caps. For the decoders, they are obtained while
7480 parsing the key management from the client.
7481 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7483 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7485 * tests/check/gst/stream.c:
7486 stream tests: Make sure we get right multicast address from stream
7487 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7489 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7491 * gst/rtsp-server/rtsp-client.c:
7492 client: ref the context until rtsp watch is alive
7493 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7495 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7497 * gst/rtsp-server/rtsp-client.c:
7498 client: Destroy the rtsp watch after connection close
7500 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7502 * gst/rtsp-server/rtsp-media.c:
7503 media: fix confusing comment
7505 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7507 * gst/rtsp-server/rtsp-session.c:
7508 rtsp-session: Timeout in header.
7509 Adding the possbilty to always have timout in header.
7510 This is configurabe with setting "timeout-always-visible".
7511 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7513 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7518 === release 1.3.2 ===
7520 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7527 * gst-rtsp-server.doap:
7530 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7533 Automatic update of common submodule
7534 From 211fa5f to 1f5d3c3
7536 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7538 * gst/rtsp-server/rtsp-client.c:
7539 client: store TCP ports in transport
7540 Store the TCP ports in the transport when we are doing RTSP over TCP.
7541 This way, we can easily get to the ports from the transport.
7542 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7544 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7546 * gst/rtsp-server/rtsp-stream.c:
7547 stream: add signals for new RTP/RTCP encoders
7548 New signals to allow the user to configure the dynamically created
7550 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7552 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7554 * gst/rtsp-server/rtsp-media.c:
7555 * gst/rtsp-server/rtsp-media.h:
7556 media: Make suspend()/unsuspend() virtual
7557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7559 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7561 * gst/rtsp-server/rtsp-client.c:
7562 client: fix send-message signal marshaller
7563 Use generic marshalling for the send-message signal. It has
7564 two POINTER arguments, not just one.
7565 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7567 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7569 * tests/check/gst/media.c:
7570 tests: add and remove pads only once
7571 In this test we simulate a dynamic pad by watching the caps event.
7572 Because of renegotiation in the base payloader now, this caps is sent
7573 multiple times but we can only deal with 1 invocation, use a variable to
7574 only 'add and remove' the pad once.
7576 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7578 * tests/check/gst/rtspserver.c:
7579 tests: add unit test for correct handling of Require headers
7580 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7582 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7584 * gst/rtsp-server/rtsp-client.c:
7585 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7586 Servers must handle Require headers and must report a failure
7587 if they don't handle any of the Required options, see RFC 2326,
7588 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7589 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7591 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7596 === release 1.3.1 ===
7598 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7604 * gst-rtsp-server.doap:
7607 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7610 Automatic update of common submodule
7611 From bcb1518 to 211fa5f
7613 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7618 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7620 * tests/check/gst/sessionmedia.c:
7621 tests: fix memory leak in sessionmedia unit test
7623 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7625 * gst/rtsp-server/rtsp-client.c:
7626 client: emit a signal before sending a message
7627 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7629 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7631 * gst/rtsp-server/rtsp-client.c:
7632 client: pass context to send_message
7633 Pass the current context to send_message, we will need it later.
7635 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7637 * gst/rtsp-server/rtsp-client.c:
7638 client: fix typo in comment
7640 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7642 * gst/rtsp-server/rtsp-media.c:
7643 media: Do not stop thread twice if default_prepare() fails
7645 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7647 * gst/rtsp-server/rtsp-client.c:
7648 client: set the watch to flushing before going to NULL
7649 First set the watch to flushing so that we unblock any current and
7650 future attempt to send data on the watch, Then set the pipeline to
7652 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7654 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7656 * gst/rtsp-server/rtsp-session-pool.c:
7657 * tests/check/gst/sessionpool.c:
7658 rtsp-session-pool: Fixes annotation
7659 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7660 in the sessionpool test.
7661 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7663 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7665 * gst/rtsp-server/rtsp-media.c:
7666 * gst/rtsp-server/rtsp-media.h:
7667 media: make media_prepare virtual
7668 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7670 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7672 * gst/rtsp-server/rtsp-media.c:
7673 * tests/check/gst/media.c:
7674 media: stop the thread in more error cases
7676 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7678 * gst/rtsp-server/rtsp-media.c:
7679 * tests/check/gst/media.c:
7680 media: allow NULL as the thread
7681 Use the default context whan passing a NULL thread.
7683 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7685 * gst/rtsp-server/rtsp-client.c:
7686 rtsp-client: indent cleanup
7687 Coverity was moaning about unreachable code, and I think it was just
7688 confused by { being before the label. We'll see if it pops up again.
7691 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7693 * gst/rtsp-server/rtsp-client.c:
7694 * gst/rtsp-server/rtsp-media.c:
7695 client: Add drop-backlog property
7696 When we have too many messages queued for a client (currently hardcoded
7697 to 100) we overflow and drop the messages. Add a drop-backlog property
7698 to control this behaviour. Setting this property to FALSE will retry
7699 to send the messages to the client by waiting for more room in the
7701 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7703 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7705 * gst/rtsp-server/rtsp-client.c:
7706 client: support for POST before GET when setting up a tunnel
7708 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7710 * gst/rtsp-server/rtsp-client.c:
7711 client: remove watch of the second client after http tunnel setup
7712 The second client will be freed after the HTTP tunnel has been set up.
7713 Make sure it's RTSP watch is never dispatched again.
7714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7716 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7718 * gst/rtsp-server/rtsp-media.c:
7719 * tests/check/gst/media.c:
7720 media: Make media_prepare() fail if port allocation fails
7721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7723 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7725 * tests/check/gst/media.c:
7726 media test: cleanup the thread pool in tests
7728 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7730 * gst/rtsp-server/rtsp-media.c:
7731 * tests/check/gst/media.c:
7732 rtsp-media: Unblock blocked streams in unprepare
7733 The streams will be blocked when a live media is prepared.
7734 The streams should be unblocked in gst_rtsp_media_unprepare.
7735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7737 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7739 * gst/rtsp-server/rtsp-media.c:
7740 media: release the state lock when going to NULL
7741 Set our state to UNPREPARING and release the state-lock before
7742 setting the pipeline to the NULL state. This way, any pad-added
7743 callback will be able to take the state-lock and check that we are now
7744 unpreparing instead of deadlocking.
7745 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7747 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7749 * gst/rtsp-server/rtsp-media.c:
7750 media: protect status with lock
7751 Make sure we only update the status with the lock.
7753 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7755 * gst/rtsp-server/rtsp-client.c:
7756 * gst/rtsp-server/rtsp-sdp.c:
7757 rtsp: update for MIKEY API changes
7759 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7761 * gst/rtsp-server/rtsp-client.c:
7762 client: parse the mikey response from the client
7763 Parse the mikey response from the client and update the policy for
7766 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7768 * gst/rtsp-server/rtsp-stream.c:
7769 * gst/rtsp-server/rtsp-stream.h:
7770 stream: add method to set crypto info
7771 Make a method to configure the crypto information of a stream.
7772 Set udpsrc in READY instead of PAUSED so that we can configure caps
7775 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7777 * gst/rtsp-server/rtsp-client.c:
7778 client: cleanup error paths
7780 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7782 * gst/rtsp-server/rtsp-media.c:
7785 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7787 * examples/test-video.c:
7788 test: enable SRTP only on RTSPS
7789 We only want to enable SRTP when doing rtsp over TLS so that we can
7790 exchange the keys in a secure way.
7792 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7794 * examples/test-video.c:
7795 test: print an error on failure
7797 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7800 * examples/test-video.c:
7801 * gst/rtsp-server/rtsp-sdp.c:
7802 * gst/rtsp-server/rtsp-stream.c:
7803 * tests/check/Makefile.am:
7804 stream: add SRTP support
7805 Install srtp encoder and decoder elements in rtpbin
7808 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7810 * tests/check/Makefile.am:
7811 * tests/check/gst/sessionpool.c:
7812 tests: Add unit tests for sessionpool
7813 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7815 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7817 * tests/check/gst/threadpool.c:
7818 tests: Improve code coverage of rtsp-threadpool tests
7819 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7821 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7823 * tests/check/gst/sessionmedia.c:
7824 tests: Improve code coverage for rtsp-session-media
7825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7827 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7829 gobject-introspection: Add annotations to support language bindings
7830 In addition a few cosmetic changes:
7831 * Adjust the order of arguments
7832 * Fix typo: occured -> occurred
7833 * Fix indentation after Return:-clauses
7834 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7836 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7838 * gst/rtsp-server/rtsp-stream.c:
7839 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7840 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7842 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7844 * gst/rtsp-server/rtsp-stream.c:
7845 stream: take caps after the session manager
7846 Take the caps for the SDP after they leave the rtpbin so that we can
7847 also get the properties added by rtpbin elements.
7849 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7851 * gst/rtsp-server/rtsp-stream.c:
7852 stream: release lock while pushing out packets
7853 Keep a cache of the transports and use this to iterate the transport
7854 while pushing packets. This allows us to release the lock early.
7855 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7857 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7859 * gst/rtsp-server/rtsp-client.c:
7860 * gst/rtsp-server/rtsp-client.h:
7861 rtsp-client: vmethod for modifying tunnel GET response
7862 Add a vmethod tunnel_http_response where the response to the HTTP GET
7863 for tunneled connections can be modified.
7864 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7866 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7868 * gst/rtsp-server/rtsp-sdp.c:
7869 sdp: make 1 media line per profile
7870 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7871 line in the SDP for each profile. The client is then supposed to pick
7872 one of the profiles in the SETUP request. Because the m= lines have the
7873 same pt, the client also knows that only 1 option is possible.
7875 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7877 * gst/rtsp-server/rtsp-media-factory.c:
7878 * gst/rtsp-server/rtsp-media-factory.h:
7879 * gst/rtsp-server/rtsp-media.c:
7880 factory: add profile property and pass to media and streams
7882 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7884 * examples/test-multicast.c:
7885 * gst/rtsp-server/rtsp-sdp.c:
7886 sdp: pass multicast connection for multicast-only stream
7887 Pass the multicast address of the stream in the connection info in the
7888 SDP so that clients try a multicast connection first.
7889 Only allow multicast connections in the test-multicast example. Also
7890 increase the TTL a little.
7892 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7895 .gitignore: Ignore gcov intermediate files
7896 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7898 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7900 * gst/rtsp-server/rtsp-stream.c:
7901 stream: release some locks in error cases
7903 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7905 docs: Enable and fix gtk-doc warnings
7906 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7907 * addresspool/mediafactory: Add missing annotation colon
7908 * stream: Annotate return value
7909 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7911 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7914 Automatic update of common submodule
7915 From fe1672e to bcb1518
7917 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7920 Automatic update of common submodule
7921 From 1a07da9 to fe1672e
7923 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7925 * examples/Makefile.am:
7926 examples: use LDADD for libs instead of LDFLAGS
7928 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7931 configure: make sure releases are in .doap file
7933 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7935 * examples/test-cgroups.c:
7936 examples: test-cgroups: don't put code with side effects into g_assert()
7937 The g_assert() might get compiled out with the right
7938 compiler/preprocessor flags.
7940 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7942 * examples/.gitignore:
7943 examples: add cgroup test binary to .gitignore
7945 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7947 * examples/test-cgroups.c:
7948 examples: fix cgroup test build
7949 Fixes build failure caused by compiler warning:
7950 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7952 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7955 .gitignore: ignore temp files created in the course of 'make check'
7957 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7959 * gst/rtsp-server/rtsp-media.c:
7960 rtsp-media: don't loose frames handling new PLAY request
7961 If client supplied a range check if the range specifies the start point.
7962 If not, then do an accurate seek to the current position. If a start
7963 point was specified do do a key unit seek to make sure the streaming
7964 starts with decodeable frames.
7965 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7967 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7969 * gst/rtsp-server/rtsp-media.c:
7970 Revert "media: only flush when setting a new start position"
7971 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7972 We need to do the flush in all cases, demuxer block currently for
7975 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7977 * gst/rtsp-server/rtsp-media.c:
7978 media: only flush when setting a new start position
7979 Only flush the pipeline when we change the start position with
7981 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7983 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7985 * gst/rtsp-server/rtsp-stream.c:
7986 stream: set ttl-mc before adding the socket
7987 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7988 never be set on socket.
7989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7991 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7993 * gst/rtsp-server/rtsp-media.c:
7994 media: stop thread if media is already prepared
7995 in gst_rtsp_media_prepare() the thread is not used if media is already
7996 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7998 https://bugzilla.gnome.org/show_bug.cgi?id=724182
8000 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
8003 build: Ship gst-rtsp-server.doap file
8005 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
8007 * tests/check/gst/rtspserver.c:
8008 tests: Fix another compiler warning with gcc
8010 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
8012 * gst/rtsp-server/rtsp-client.c:
8013 * gst/rtsp-server/rtsp-mount-points.c:
8014 * gst/rtsp-server/rtsp-stream.c:
8015 * tests/check/gst/client.c:
8016 rtsp-server: Fix lots of compiler warnings with clang
8018 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
8021 * gst-rtsp-server.doap:
8022 * tests/Makefile.am:
8023 configure: Synchronise with the configure scripts of the other modules
8025 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
8028 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
8030 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
8032 * gst/rtsp-server/rtsp-media.c:
8033 * gst/rtsp-server/rtsp-stream.c:
8034 Revert "rtsp-server: support build against last stable release"
8035 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
8036 Let us require 1.2.3 now, which is going to be released in a few
8039 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
8041 * gst/rtsp-server/rtsp-session-media.c:
8042 * gst/rtsp-server/rtsp-stream-transport.c:
8043 session: improve RTP-Info
8044 Ignore streams that can't generate RTP-Info instead of failing.
8045 Don't return the empty string when all streams are unconfigured but
8046 return NULL so that we don't generate and empty RTP-Info header.
8047 Improve docs a little.
8049 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
8051 * gst/rtsp-server/rtsp-session-media.c:
8052 Don't free rtpinfo GString when it is NULL
8053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8055 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
8057 * gst/rtsp-server/rtsp-media.c:
8058 media: only set keyframe flag when modifying start
8059 Only set the keyframe flag when we modify the start position. The
8060 keyframe flag should probably be ignored when no change is requested but
8061 until we can claim this is all documented properly and all demuxer
8062 implement this, avoid setting the flag.
8063 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
8065 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
8067 * gst/rtsp-server/rtsp-thread-pool.c:
8068 thread-pool: Unref source after mainloop has quit to avoid races in GLib
8069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
8071 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
8073 * gst/rtsp-server/rtsp-stream.c:
8074 stream: handle NULL seqnum and rtptime arguments
8076 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
8078 * gst/rtsp-server/rtsp-thread-pool.c:
8079 * tests/check/gst/threadpool.c:
8080 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
8081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
8083 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
8085 * gst/rtsp-server/rtsp-stream.c:
8086 stream: add fallback for missing stats property
8087 Use a fallback when the payloader does not have a stats property
8088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8090 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
8093 Automatic update of common submodule
8094 From f7bc1c3 to 1a07da9
8096 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
8098 * gst/rtsp-server/rtsp-stream.c:
8099 stream: don't leak stats structure
8100 Don't leak the stats structure and deal with NULL stats.
8102 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
8104 * gst/rtsp-server/rtsp-stream.c:
8105 stream: Get rtpinfo properties atomically from payloader
8106 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
8108 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
8110 * gst/rtsp-server/rtsp-media.c:
8111 media: refactor state change functions and signals
8112 Make functions to set the target state and the pipeline state and emit
8113 the signals from those functions.
8115 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
8117 * gst/rtsp-server/rtsp-media.c:
8118 * gst/rtsp-server/rtsp-media.h:
8119 media: add signal to notify of pending state changes
8121 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
8123 * gst/rtsp-server/rtsp-media.c:
8124 * gst/rtsp-server/rtsp-stream.c:
8125 rtsp-server: support build against last stable release
8126 Until 1.2.3 is out with the new get_type function and we
8129 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
8131 * gst/rtsp-server/rtsp-stream.c:
8132 stream: fix compilation
8134 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
8136 * gst/rtsp-server/rtsp-media.c:
8137 * gst/rtsp-server/rtsp-media.h:
8138 * gst/rtsp-server/rtsp-stream.c:
8139 * gst/rtsp-server/rtsp-stream.h:
8140 stream: add property to configure profiles
8142 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
8144 * gst/rtsp-server/rtsp-client.c:
8145 client: let stream check supported transport
8146 Delegate the check if a transport is allowed to the stream.
8147 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
8149 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
8151 * gst/rtsp-server/rtsp-stream.c:
8152 * gst/rtsp-server/rtsp-stream.h:
8153 stream: add method to check supported transport
8154 Add a method to check if a transport is supported
8156 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
8159 configure.ac: Only check for gstreamer-check, not check
8160 We include check in gstreamer-check since quite some time now.
8162 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
8164 * gst/rtsp-server/rtsp-session-media.c:
8165 * gst/rtsp-server/rtsp-stream-transport.c:
8166 * gst/rtsp-server/rtsp-stream.c:
8167 * gst/rtsp-server/rtsp-stream.h:
8168 stream: return clock-rate from get_rtpinfo
8169 And use it to correct the rtptime to the requested start-time.
8170 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
8172 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
8174 * gst/rtsp-server/rtsp-session-media.c:
8175 * gst/rtsp-server/rtsp-stream-transport.c:
8176 * gst/rtsp-server/rtsp-stream-transport.h:
8177 session-media: calculate start-time
8179 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
8181 * gst/rtsp-server/rtsp-stream-transport.c:
8182 * gst/rtsp-server/rtsp-stream.c:
8183 * gst/rtsp-server/rtsp-stream.h:
8184 stream: also return the running-time
8185 Return the running-time in the rtpinfo as well.
8187 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
8189 * gst/rtsp-server/rtsp-client.c:
8190 * gst/rtsp-server/rtsp-session-media.c:
8191 * gst/rtsp-server/rtsp-session-media.h:
8192 * gst/rtsp-server/rtsp-stream-transport.c:
8193 * gst/rtsp-server/rtsp-stream-transport.h:
8194 session-media: let the session-media make the RTPInfo
8195 Add method to create the RTPInfo for a stream-transport.
8196 Add method to create the RTPInfo for all stream-transports in a
8198 Use the session-media RTPInfo code in client. This allows us to refactor
8199 another method to link the TCP callbacks.
8201 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8203 mount-points: sort sequence before g_sequence_lookup
8204 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
8205 sort sequence if dirty, otherwise lookup will fail.
8206 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
8208 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
8211 configure: rename package from gst-rtsp to gst-rtsp-server
8212 To match git module name and avoid confusion with the
8213 rtsp lib in gst-plugins-base and rtsp plugin in -good.
8215 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
8218 configure: bump core/base/good requirement to 1.2.0
8219 Bump to released stable version and make implicit
8220 requirements explicit.
8222 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
8227 Fix broken gettext setup which is not used anyway
8229 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
8232 Automatic update of common submodule
8233 From dbedaa0 to d48bed3
8235 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
8237 * gst/rtsp-server/rtsp-client.c:
8238 * gst/rtsp-server/rtsp-media.c:
8239 * gst/rtsp-server/rtsp-media.h:
8240 media: add setup_sdp vmethod
8241 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
8242 gst_rtsp_media_setup_sdp.
8243 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
8245 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
8247 * gst/rtsp-server/rtsp-stream.c:
8248 rtsp-stream: Check return value of sscanf
8249 streamid is only valid if sscanf matched something.
8251 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
8253 * gst/rtsp-server/rtsp-client.c:
8254 rtsp-client: Fix iteration
8255 Wouldn't even enter the code block otherwise (i++ was used as the check
8256 and not the postfix).
8258 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
8260 * gst/rtsp-server/rtsp-client.c:
8261 * gst/rtsp-server/rtsp-client.h:
8262 client: add vmethod to configure media and streams
8263 Implement a vmethod that can be used to configure the media and the
8264 streams based on the current context. Handle the blocksize handling in
8265 the default handler.
8266 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
8268 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8271 Make git ignore more unit test binaries
8273 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8275 * gst/rtsp-server/rtsp-address-pool.h:
8276 * gst/rtsp-server/rtsp-auth.h:
8277 * gst/rtsp-server/rtsp-client.h:
8278 * gst/rtsp-server/rtsp-context.h:
8279 * gst/rtsp-server/rtsp-media-factory-uri.h:
8280 * gst/rtsp-server/rtsp-media-factory.h:
8281 * gst/rtsp-server/rtsp-media.h:
8282 * gst/rtsp-server/rtsp-mount-points.h:
8283 * gst/rtsp-server/rtsp-server.h:
8284 * gst/rtsp-server/rtsp-session-media.h:
8285 * gst/rtsp-server/rtsp-session-pool.h:
8286 * gst/rtsp-server/rtsp-session.h:
8287 * gst/rtsp-server/rtsp-stream-transport.h:
8288 * gst/rtsp-server/rtsp-stream.h:
8289 * gst/rtsp-server/rtsp-thread-pool.h:
8290 * gst/rtsp-server/rtsp-token.h:
8291 rtsp-server: add padding to many public structures
8292 Not mini objects though, since they are not subclassable
8293 anyway, nor kept on the stack or inlined in a structure.
8295 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8297 media: add new create_rtpbin vmethod
8298 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8299 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8301 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8303 * tests/check/gst/media.c:
8304 tests: fix memory leak, free test's thread pool
8305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8307 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8309 * gst/rtsp-server/rtsp-stream-transport.c:
8310 stream-transport: free url in finalize
8312 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8314 * gst/rtsp-server/rtsp-media.c:
8315 media: also do state change in suspended state
8317 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8319 * gst/rtsp-server/rtsp-client.c:
8320 * gst/rtsp-server/rtsp-media.c:
8321 media: also handle prepare and range in suspended state
8322 When we are suspended, we are already prepared.
8323 We can get the range in the suspended state.
8325 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8327 * tests/check/Makefile.am:
8328 * tests/check/gst/sessionmedia.c:
8329 check: add test for uri in setup
8330 Added unit tests for the new functionality in GstRTSPStreamTransport.
8331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8333 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8335 * gst/rtsp-server/rtsp-client.c:
8336 client: store setup uri and use in PLAY response
8337 Store the uri used when doing the setup and use that in the PLAY
8339 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8341 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8343 * gst/rtsp-server/rtsp-stream-transport.c:
8344 * gst/rtsp-server/rtsp-stream-transport.h:
8345 stream-transport: add method to get/set url
8347 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8349 * gst/rtsp-server/rtsp-client.c:
8350 client: suspend after SDP and unsuspend before PLAYING
8351 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8354 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8356 * gst/rtsp-server/rtsp-media-factory.c:
8357 * gst/rtsp-server/rtsp-media-factory.h:
8358 * gst/rtsp-server/rtsp-media.c:
8359 * gst/rtsp-server/rtsp-media.h:
8360 * gst/rtsp-server/rtsp-session-media.c:
8361 * gst/rtsp-server/rtsp-session.c:
8362 * tests/check/gst/media.c:
8363 * tests/check/gst/mediafactory.c:
8364 media: add suspend modes
8365 Add support for different suspend modes. The stream is suspended right after
8366 producing the SDP and after PAUSE. Different suspend modes are available that
8367 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8368 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8369 state and RESET will bring the pipeline to the NULL state.
8370 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8371 this means that the pipeline needs to be prerolled again.
8372 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8373 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8375 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8377 * gst/rtsp-server/rtsp-media.c:
8378 media: start live streams in blocked state
8379 Start live streams in the blocked state and make them preroll using the
8380 messages. This ensure that no data is played by the sink until we explicitly
8381 unblock the stream right before going to PLAYING.
8382 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8384 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8386 * gst/rtsp-server/rtsp-media.c:
8387 media: refactor starting and waiting for preroll
8388 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8389 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8391 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8393 * gst/rtsp-server/rtsp-stream.c:
8394 * gst/rtsp-server/rtsp-stream.h:
8395 stream: add API to block streams
8396 Add an API to block on the streams and make it post a message.
8397 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8398 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8400 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8402 * docs/libs/Makefile.am:
8403 docs: Specify the override file
8404 Even if it's empty (for now) it avoids make distcheck complaining
8406 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8408 * gst/rtsp-server/rtsp-media.c:
8409 media: move default implementations to where they are used
8411 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8413 * gst/rtsp-server/rtsp-media.c:
8414 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8415 We need to take the state_lock when calling this method.
8417 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8419 * gst/rtsp-server/rtsp-media.c:
8420 media: handle add-added on non-bins too
8421 Handle dynamic payloaders that are not bins, as used in the unit-test.
8423 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8425 * gst/rtsp-server/rtsp-media-factory.c:
8426 * gst/rtsp-server/rtsp-media-factory.h:
8427 * gst/rtsp-server/rtsp-media.c:
8428 rtsp-media/-factory: Fix request pad name comments
8429 These must be escaped for gtk-doc to parse the comments without warnings.
8431 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8433 rtsp-media: remove transports if media is in error status
8434 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8435 trying to change to GST_STATE_NULL and media is in error status, we
8436 remove all transports.
8437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8439 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8441 * gst/rtsp-server/rtsp-media.c:
8442 rtsp-media: use element metadata to find payloader
8443 Use the element metadata to find the payloader instead of checking
8445 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8447 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8449 rtsp-stream: add getter for payload type
8450 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8451 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8452 element and create the stream with this one instead of the dynpay%d
8454 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8456 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8458 * gst/rtsp-server/rtsp-client.c:
8459 * gst/rtsp-server/rtsp-context.h:
8460 * gst/rtsp-server/rtsp-media.c:
8461 * gst/rtsp-server/rtsp-mount-points.c:
8462 * gst/rtsp-server/rtsp-server.c:
8463 * gst/rtsp-server/rtsp-token.c:
8464 rtsp-*: Refer to NULL as a constant in comments
8466 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8468 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8470 rtsp-*: Fix type name typos in comments
8471 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8472 * rtsp-auth: Refer to part of constant name as text
8473 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8474 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8475 * rtsp-stream: Fix typo when refering to GstBin
8476 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8478 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8481 * docs/libs/gst-rtsp-server-docs.sgml:
8482 * docs/libs/gst-rtsp-server-sections.txt:
8483 docs: Improve documentation
8484 * Include annotation-glossary to quiet gtk-doc
8485 * Rename remaining ClientState -> Context
8486 * Rename object hierarchy file
8487 * Remove stale chapter references
8488 * Add missing function and object references
8489 * Include missing GstRTSPAddressPoolResult
8490 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8492 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8494 * gst/rtsp-server/rtsp-client.c:
8495 * gst/rtsp-server/rtsp-server.c:
8496 * gst/rtsp-server/rtsp-session-pool.c:
8497 * gst/rtsp-server/rtsp-session.c:
8498 * gst/rtsp-server/rtsp-stream.c:
8499 rtsp-server: sprinkle some allow-none annotations for g-i
8501 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8503 * gst/rtsp-server/rtsp-stream.c:
8504 * gst/rtsp-server/rtsp-stream.h:
8505 stream: add method to filter transports
8506 Add a method to safely iterate and collect the stream transports
8507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8509 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8511 * gst/rtsp-server/rtsp-client.c:
8512 * gst/rtsp-server/rtsp-server.c:
8513 * gst/rtsp-server/rtsp-session-pool.c:
8514 * gst/rtsp-server/rtsp-session.c:
8515 rtsp: allow NULL func in filters
8516 Passing a null function make the filters return a list of
8519 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8521 * gst/rtsp-server/rtsp-address-pool.c:
8522 * tests/check/gst/addresspool.c:
8523 address-pool: fix address increment
8524 Use a guint instead of guint8 to increment the address. It's still not
8525 completely correct because a guint might not be able to hold the complete
8526 address range, but that's an enhacement for later.
8527 Add unit test to test improved behaviour.
8528 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8530 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8532 * gst/rtsp-server/rtsp-client.c:
8533 * tests/check/gst/client.c:
8534 client: allow absolute path in requests
8535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8537 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8539 * gst/rtsp-server/rtsp-client.c:
8540 * gst/rtsp-server/rtsp-client.h:
8541 client: make make_path_from_uri a vmethod
8543 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8545 * docs/libs/gst-rtsp-server-sections.txt:
8546 * gst/rtsp-server/rtsp-stream.c:
8547 * gst/rtsp-server/rtsp-stream.h:
8548 * tests/check/Makefile.am:
8549 * tests/check/gst/stream.c:
8550 stream: Add functions to get rtp and rtcp sockets
8551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8553 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8555 * gst/rtsp-server/rtsp-context.c:
8556 * gst/rtsp-server/rtsp-context.h:
8557 context: defing a GType for the context
8558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8560 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8562 * gst/rtsp-server/Makefile.am:
8563 * gst/rtsp-server/rtsp-auth.c:
8564 * gst/rtsp-server/rtsp-context.c:
8565 * gst/rtsp-server/rtsp-media.c:
8566 * gst/rtsp-server/rtsp-mount-points.c:
8567 * gst/rtsp-server/rtsp-server.h:
8568 * gst/rtsp-server/rtsp-session-media.c:
8569 * gst/rtsp-server/rtsp-session.c:
8570 * gst/rtsp-server/rtsp-stream.c:
8571 Fixed several GIR warnings
8573 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8575 * gst/rtsp-server/rtsp-auth.c:
8578 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8580 * tests/check/Makefile.am:
8581 * tests/check/gst/token.c:
8582 tests: Add unit tests for token
8583 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8585 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8587 * gst/rtsp-server/rtsp-token.c:
8588 token: Validate args for gst_rtsp_token_is_allowed
8589 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8591 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8593 * gst/rtsp-server/rtsp-token.c:
8594 token: Fix bug when creating empty token
8595 We always want to have a valid GstStructure in the token.
8596 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8598 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8600 * gst/rtsp-server/rtsp-thread-pool.c:
8601 thread-pool: avoid race in shutdown
8602 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8603 don't actually stop the mainloop ever. Solve this race by adding an idle source
8604 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8605 if quit was called before we started it.
8607 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8609 * tests/check/Makefile.am:
8610 * tests/check/gst/permissions.c:
8611 tests: Add unit tests for permissions
8612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8614 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8616 * tests/check/gst/mediafactory.c:
8617 tests: Test mediafactory permissions
8618 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8620 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8622 * gst/rtsp-server/rtsp-permissions.c:
8623 permissions: Fix refcounting when adding/removing roles
8624 Previously a role that was removed was unreffed twice, and when
8625 replacing an existing role the replaced role was freed while still being
8626 referenced. Both bugs are now fixed.
8627 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8629 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8631 * tests/check/gst/media.c:
8632 * tests/check/gst/mediafactory.c:
8633 * tests/check/gst/rtspserver.c:
8634 tests: Check gst_rtsp_url_parse return value
8635 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8637 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8640 Automatic update of common submodule
8641 From 865aa20 to dbedaa0
8643 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8645 * gst/rtsp-server/rtsp-server.c:
8646 rtsp-server: Fix socket leak
8647 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8649 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8651 * gst/rtsp-server/rtsp-session-pool.c:
8652 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8653 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8655 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8657 * examples/.gitignore:
8658 * examples/test-video.c:
8659 examples: fix compilation when WITH_AUTH is defined
8660 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8662 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8665 gitignore: Add new test binary
8667 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8669 * tests/check/Makefile.am:
8670 * tests/check/gst/threadpool.c:
8671 thread-pool: Add unit test for the thread pools
8672 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8674 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8676 * gst/rtsp-server/rtsp-thread-pool.c:
8677 thread-pool: Fix thread leak when reusing threads
8678 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8680 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8682 * gst/rtsp-server/rtsp-server.c:
8683 * tests/check/gst/rtspserver.c:
8684 tests: fixed racy behavior in rtspserver tests
8685 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8687 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8689 * tests/check/gst/addresspool.c:
8690 tests: Improve address pool unit tests
8691 Add a range with mixed IPV4 and IPV6 addresses to pool.
8692 Get an IPV4 address from an IPV6-only pool.
8693 Get an IPV6 address from an IPV4-only pool.
8694 Reserve a IPV6 address from an IPV4-only pool.
8695 Check for unicast addresses in multicast-only pool.
8696 Check for unicast addresses in uni-/multicast-mixed pool.
8697 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8699 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8701 * gst/rtsp-server/rtsp-client.c:
8702 client: append query string in PAUSE/PLAY/TEARDOWN as well
8704 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8706 * gst/rtsp-server/rtsp-client.c:
8707 client: Add query to control path
8708 If the SETUP url contains a query it must be appended to the control
8709 path so that it matches any already created stream in the media. The
8710 query will also be appended to the session media path.
8712 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8714 * gst/rtsp-server/rtsp-media.c:
8715 rtsp-media: remove old line
8717 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8719 * gst/rtsp-server/rtsp-stream.c:
8720 stream: Correct control comparison
8721 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8723 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8725 * gst/rtsp-server/rtsp-media.c:
8726 media: Check dynamically if the pipeline supports seeking
8727 We should not depend on whether or not the pipeline state change
8728 returned NO_PREROLL or not. A media could dynamically change its
8729 element and switch from seekable to non seekable so it's best to test
8730 the seekable nature of the pipeline dynamically when we try to do a seek.
8732 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8734 * gst/rtsp-server/rtsp-media.c:
8735 media: Return FALSE if seeking is not supported
8737 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8739 * gst/rtsp-server/rtsp-media.c:
8740 rtsp-media: don't seek accurate by default
8741 Accurate seeking is perhaps a little overkill in the most common situation and
8742 causes some formats (mp3) over slow media to seek extremely slowly.
8744 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8746 * tests/check/gst/rtspserver.c:
8747 tests: fix unit test
8748 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8750 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8752 * gst/rtsp-server/rtsp-client.c:
8753 client: Reply 400 if media cannot be constructed
8754 Reply 400 Bad Request instead of 503 Service Unavailable if media
8755 cannot be constructed in SETUP.
8756 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8758 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8760 * gst/rtsp-server/rtsp-client.c:
8761 client: Send setup reply once only
8762 If find_media() failed in handle_setup_request() two replies was sent.
8763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8765 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8768 Automatic update of common submodule
8769 From 6b03ba7 to 865aa20
8771 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8773 * gst/rtsp-server/rtsp-server.c:
8774 server: Emit client-connected signal earlier
8775 Emit client-connected before the client ref is given to a GSource,
8776 otherwise client-connected can be emitted after the client object has
8779 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8781 * gst/rtsp-server/rtsp-address-pool.c:
8782 * gst/rtsp-server/rtsp-address-pool.h:
8783 * gst/rtsp-server/rtsp-stream.c:
8784 * tests/check/gst/addresspool.c:
8785 addresspool: return reason of failure
8786 Let gst_rtsp_address_pool_reserve_address() return the reason why
8787 the address could not be reserved.
8788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8790 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8793 autogen.sh: Sync behaviour with other GStreamer modules
8794 Allows building from outside of tree amongst other things
8796 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8799 Automatic update of common submodule
8800 From b613661 to 6b03ba7
8802 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8805 Automatic update of common submodule
8806 From 74a6857 to b613661
8808 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8811 Automatic update of common submodule
8812 From 01a7a46 to 74a6857
8814 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8816 * gst/rtsp-server/rtsp-client.c:
8817 client: Do not read beyond end of path string
8818 If the setup was done without a control url, make sure we don't try to read the
8819 non-existing control string and crash.
8821 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8823 * gst/rtsp-server/rtsp-client.c:
8824 client: Fix RTPInfo header
8825 Refactor the method to make the content_base.
8826 Use the content-base and the control url to construct the RTPInfo
8829 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8831 * gst/rtsp-server/rtsp-client.c:
8832 client: map url to path only in describe
8833 Only map the request url to a path in the DESCRIBE method. The SDP then
8834 contains the base and control urls that should be used to SETUP/PAUSE/
8835 PLAY/TEARDOWN the media.
8837 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8839 * gst/rtsp-server/rtsp-client.c:
8840 Revert "client: map URL to path in requests"
8841 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8842 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8843 contains the base and control urls which are used in the SETUP, PLAY,
8844 PAUSE and TEARDOWN requests.
8846 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8848 * gst/rtsp-server/rtsp-client.c:
8849 client: map URL to path in requests
8851 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8853 * gst/rtsp-server/rtsp-client.c:
8854 * gst/rtsp-server/rtsp-mount-points.c:
8855 * gst/rtsp-server/rtsp-mount-points.h:
8856 mount-points: make vmethod to make path from uri
8857 Make a vmethod to transform an url into a path. The path is then used to lookup
8858 the factory. This makes it possible to also use other bits of the url, such as
8859 the query parameters, to locate the factory.
8861 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8863 * gst/rtsp-server/rtsp-thread-pool.c:
8864 * gst/rtsp-server/rtsp-thread-pool.h:
8865 thread-pool: Add cleanup to wait for the threadpool to finish
8866 Also fix race condition if two threads are asking for the first
8867 thread from the thread pool at once. This would case two internal
8868 GThreadPools to be created.
8869 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8871 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8873 * gst/rtsp-server/rtsp-client.c:
8874 * tests/check/gst/client.c:
8875 client: free threadpool
8876 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8878 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8880 * tests/check/gst/mountpoints.c:
8881 mountpoints tests: unref matched factories
8882 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8884 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8886 * tests/check/gst/media.c:
8887 media tests: unref thread pool and caps
8888 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8890 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8892 * gst/rtsp-server/rtsp-auth.c:
8893 * gst/rtsp-server/rtsp-media-factory.c:
8894 * gst/rtsp-server/rtsp-media.c:
8895 auth, media, media-factory: unref permissions
8896 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8898 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8900 * examples/Makefile.am:
8901 Makefile: add rule for appsrc example
8903 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8905 * examples/test-appsrc.c:
8906 tests: add appsrc example
8907 Add an example on how to use appsrc to feed the server pipeline with data.
8909 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8911 * gst/rtsp-server/rtsp-client.c:
8912 rtsp-client: remove query part from content-base string
8913 Make sure that after the control url has been resolved, it's
8914 not a part of the query-string.
8915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8917 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-client.c:
8920 client: don't check url in response
8921 There is no url or method in the response to check
8923 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8925 * gst/rtsp-server/rtsp-client.c:
8926 * gst/rtsp-server/rtsp-client.h:
8927 Add handle-response signal for when we receive a GET_PARAMETER response
8929 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8931 * gst/rtsp-server/rtsp-server.c:
8932 Fix gst_rtsp_server_client_filter, using wrong variable type
8934 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8936 * gst/rtsp-server/rtsp-media-factory-uri.c:
8937 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8938 For AAC we need to check for framed=true instead of parsed=true.
8939 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8941 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8943 * gst/rtsp-server/rtsp-stream.c:
8944 stream: optimize pipeline for protocols
8945 When TCP is not an allowed protocol for the stream, avoid creating the
8946 appsrc/appsink/queue and tee elements.
8948 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8950 * gst/rtsp-server/rtsp-media.c:
8951 media: set protocols on streams
8953 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8955 * gst/rtsp-server/rtsp-client.c:
8956 client: use protocols supported by stream
8958 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8960 * gst/rtsp-server/rtsp-media-factory.c:
8961 * gst/rtsp-server/rtsp-media.c:
8962 * gst/rtsp-server/rtsp-stream.c:
8963 media-factory: allow all protocols
8965 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8967 * gst/rtsp-server/rtsp-media.c:
8968 media: configure protocols in new streams
8970 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8972 * gst/rtsp-server/rtsp-stream.c:
8973 * gst/rtsp-server/rtsp-stream.h:
8974 stream: add protocols property
8976 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8978 * gst/rtsp-server/rtsp-media.c:
8979 rtsp-media: send state in "new-state" signal
8980 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8982 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8985 build: add subdir-objects to AM_INIT_AUTOMAKE
8986 Fixes warnings with automake 1.14
8987 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8989 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * docs/libs/gst-rtsp-server-sections.txt:
8992 * gst/rtsp-server/rtsp-client.c:
8993 * gst/rtsp-server/rtsp-server.c:
8994 * gst/rtsp-server/rtsp-server.h:
8995 server: add method to iterate clients of server
8997 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8999 * gst/rtsp-server/rtsp-media.c:
9000 * gst/rtsp-server/rtsp-media.h:
9001 Add vmethod for rtsp-media subclass to access rtpbin
9003 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9005 * gst/rtsp-server/rtsp-client.h:
9006 small documentation fix
9008 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9010 * gst/rtsp-server/rtsp-client.c:
9011 Do not take range header if range is invalid
9013 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9015 * docs/libs/gst-rtsp-server-sections.txt:
9016 * gst/rtsp-server/rtsp-media.c:
9017 media: add docs for new method
9019 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9021 * gst/rtsp-server/rtsp-media.c:
9022 * gst/rtsp-server/rtsp-media.h:
9023 Add API to rtsp-media set the pipeline's state
9025 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9027 * gst/rtsp-server/rtsp-media.c:
9028 Update current position/duration when gst_rtsp_media_get_range_string is called
9030 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9032 * examples/test-cgroups.c:
9033 tests: add some more docs
9035 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9037 * examples/test-cgroups.c:
9038 * gst/rtsp-server/Makefile.am:
9039 * gst/rtsp-server/rtsp-auth.c:
9040 * gst/rtsp-server/rtsp-auth.h:
9041 * gst/rtsp-server/rtsp-client.c:
9042 * gst/rtsp-server/rtsp-client.h:
9043 * gst/rtsp-server/rtsp-context.c:
9044 * gst/rtsp-server/rtsp-context.h:
9045 * gst/rtsp-server/rtsp-params.c:
9046 * gst/rtsp-server/rtsp-params.h:
9047 * gst/rtsp-server/rtsp-server.c:
9048 * gst/rtsp-server/rtsp-thread-pool.c:
9049 * gst/rtsp-server/rtsp-thread-pool.h:
9050 * tests/check/gst/client.c:
9051 ClientState -> Context
9052 Rename the clientstate to context and put the code in a separate file.
9054 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9056 * examples/test-auth.c:
9057 * gst/rtsp-server/rtsp-auth.c:
9058 * gst/rtsp-server/rtsp-auth.h:
9059 auth: add support for default token
9060 The default token is used when the user is not authenticated and can be used to
9061 give minimal permissions.
9063 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9065 * examples/test-auth.c:
9066 * gst/rtsp-server/rtsp-auth.c:
9067 auth: use defines when possible
9069 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9071 * gst/rtsp-server/rtsp-address-pool.c:
9072 address-pool: improve docs
9074 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9076 * gst/rtsp-server/rtsp-permissions.c:
9077 permissions: add the role to the copy
9079 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
9081 * gst/rtsp-server/rtsp-permissions.c:
9082 permissions: Also copy the roles
9084 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
9086 * gst/rtsp-server/rtsp-permissions.c:
9087 permissions: Make it build
9089 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9091 * gst/rtsp-server/rtsp-address-pool.h:
9094 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9096 * docs/libs/gst-rtsp-server-sections.txt:
9097 * gst/rtsp-server/rtsp-auth.c:
9098 * gst/rtsp-server/rtsp-auth.h:
9099 * gst/rtsp-server/rtsp-media.c:
9100 * gst/rtsp-server/rtsp-session-media.c:
9101 * gst/rtsp-server/rtsp-stream-transport.c:
9102 * gst/rtsp-server/rtsp-stream-transport.h:
9103 * gst/rtsp-server/rtsp-stream.c:
9104 * tests/check/gst/client.c:
9107 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9109 * docs/libs/gst-rtsp-server-sections.txt:
9110 * gst/rtsp-server/rtsp-address-pool.c:
9111 * gst/rtsp-server/rtsp-address-pool.h:
9112 * tests/check/gst/addresspool.c:
9113 * tests/check/gst/rtspserver.c:
9114 address-pool: cleanups
9115 Remove redundant method, improve docs.
9117 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9119 * docs/libs/gst-rtsp-server-sections.txt:
9120 * gst/rtsp-server/rtsp-auth.h:
9121 * gst/rtsp-server/rtsp-permissions.c:
9122 * gst/rtsp-server/rtsp-permissions.h:
9123 * gst/rtsp-server/rtsp-token.c:
9126 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * gst/rtsp-server/rtsp-permissions.c:
9129 permissions: implement _remove_role
9131 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9133 * gst/rtsp-server/rtsp-permissions.c:
9134 permissions: update docs
9136 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9138 * tests/check/gst/client.c:
9139 tests: simplify tests
9140 Client settings are now disabled by default so we don't need an auth
9141 module to disable them.
9143 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9145 * gst/rtsp-server/rtsp-auth.c:
9146 auth: add default authorizations
9147 When no auth module is specified, use our table of defaults to look up the
9148 default value of the check instead of always allowing everything. This was
9149 we can disallow client settings by default.
9151 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9154 README: update readme
9156 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9158 * gst/rtsp-server/rtsp-thread-pool.c:
9159 * gst/rtsp-server/rtsp-thread-pool.h:
9160 thread-pool: add more docs
9162 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9164 * gst/rtsp-server/rtsp-thread-pool.c:
9165 * gst/rtsp-server/rtsp-thread-pool.h:
9166 thread-pool: fix race in thread reuse
9167 If we try to reuse a thread right after we made it stop, we end up using a
9168 stopped thread. Catch this case and only reuse threads that are not stopping.
9170 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9172 * gst/rtsp-server/rtsp-server.c:
9173 server: add small debug
9175 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * tests/check/gst/client.c:
9179 Add some permissions to media so we can use the auth and enable
9182 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9184 * gst/rtsp-server/rtsp-client.c:
9185 client: support pushed context in handle_request
9186 If we already have a pushed state, reuse it and add our own things. This makes
9187 it easier to write tests.
9189 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9191 * gst/rtsp-server/rtsp-auth.c:
9192 auth: don't auth on methods
9193 Don't authorize on methods anymore but on the resources that we
9194 try to access, this is more flexible.
9195 Move the authorization checks to where they are needed and let the
9196 check return the response on error.
9198 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9200 * gst/rtsp-server/rtsp-mount-points.c:
9201 mount-points: add some debug
9203 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9205 * tests/check/gst/client.c:
9206 tests: almost fix test
9208 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9210 * gst/rtsp-server/rtsp-auth.c:
9211 * gst/rtsp-server/rtsp-auth.h:
9212 * gst/rtsp-server/rtsp-client.c:
9213 * gst/rtsp-server/rtsp-client.h:
9214 * gst/rtsp-server/rtsp-server.c:
9215 * gst/rtsp-server/rtsp-server.h:
9216 auth: let the auth module check client_settings
9217 Let the auth module decide if client settings are allowed for the
9220 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9222 * gst/rtsp-server/rtsp-token.c:
9223 * gst/rtsp-server/rtsp-token.h:
9224 token: add method to check boolean permission
9226 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9228 * examples/test-auth.c:
9229 * examples/test-cgroups.c:
9230 * gst/rtsp-server/rtsp-token.c:
9231 * gst/rtsp-server/rtsp-token.h:
9232 token: simplify token constructor
9233 Use variable arguments to make easier API.
9235 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9237 * examples/test-auth.c:
9238 * examples/test-cgroups.c:
9239 * gst/rtsp-server/rtsp-media-factory.c:
9240 * gst/rtsp-server/rtsp-media-factory.h:
9241 media-factory: add convenience API for factory
9243 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9245 * examples/test-auth.c:
9246 * examples/test-cgroups.c:
9247 * gst/rtsp-server/rtsp-permissions.c:
9248 * gst/rtsp-server/rtsp-permissions.h:
9249 permissions: simplify API a little
9250 Avoid passing GstStructure in the add_role method, use varargs instead
9251 to construct the structure behind the scenes. We can then also use the
9252 structure name as the role and simplify some more logic.
9254 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9256 * gst/rtsp-server/rtsp-auth.c:
9259 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9261 * gst/rtsp-server/rtsp-auth.c:
9262 * gst/rtsp-server/rtsp-auth.h:
9263 * gst/rtsp-server/rtsp-client.c:
9264 auth: handle unauthorized response
9265 Move handling of the unauthorized response to the auth module, it can add
9266 the appropriate headers to request authorization for the required method
9267 much better than the client.
9269 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9271 * gst/rtsp-server/rtsp-client.c:
9272 * gst/rtsp-server/rtsp-client.h:
9273 client: allow for sending any message, not only requests
9274 Change the _send_request() method to _send_message() so that we
9275 can both send requests and replies.
9277 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9279 * docs/libs/gst-rtsp-server-sections.txt:
9280 * gst/rtsp-server/rtsp-server.h:
9283 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9285 * examples/test-video.c:
9286 * gst/rtsp-server/rtsp-auth.c:
9287 * gst/rtsp-server/rtsp-auth.h:
9288 * gst/rtsp-server/rtsp-server.c:
9289 * gst/rtsp-server/rtsp-server.h:
9290 auth: move TLS handling to auth module
9291 Remove the TLS settings on the server and move it to the auth module because
9292 that is where security related bits go.
9294 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9296 * gst/rtsp-server/rtsp-client.c:
9297 * gst/rtsp-server/rtsp-client.h:
9298 client: add state push/pop
9300 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9302 * gst/rtsp-server/rtsp-client.c:
9303 * gst/rtsp-server/rtsp-client.h:
9304 client: add connection to state
9306 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * gst/rtsp-server/rtsp-mount-points.c:
9309 mount-points: fix debug
9311 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9313 * tests/check/gst/media.c:
9314 tests: fix media test
9316 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9318 * gst/rtsp-server/rtsp-thread-pool.c:
9319 thread-pool: we don't require a state
9321 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9323 * gst/rtsp-server/rtsp-server.c:
9324 server: let context ref the server
9325 So that we don't risk losing the server object early anc crash.
9327 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9329 * tests/check/gst/client.c:
9330 tests: fix client test
9332 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * docs/libs/gst-rtsp-server-docs.sgml:
9336 * docs/libs/gst-rtsp-server-sections.txt:
9337 * gst/rtsp-server/rtsp-address-pool.c:
9338 * gst/rtsp-server/rtsp-auth.c:
9339 * gst/rtsp-server/rtsp-client.c:
9340 * gst/rtsp-server/rtsp-client.h:
9341 * gst/rtsp-server/rtsp-media-factory-uri.c:
9342 * gst/rtsp-server/rtsp-media-factory.c:
9343 * gst/rtsp-server/rtsp-media-factory.h:
9344 * gst/rtsp-server/rtsp-media.c:
9345 * gst/rtsp-server/rtsp-mount-points.c:
9346 * gst/rtsp-server/rtsp-params.c:
9347 * gst/rtsp-server/rtsp-permissions.c:
9348 * gst/rtsp-server/rtsp-sdp.c:
9349 * gst/rtsp-server/rtsp-server.c:
9350 * gst/rtsp-server/rtsp-server.h:
9351 * gst/rtsp-server/rtsp-session-media.c:
9352 * gst/rtsp-server/rtsp-session-pool.c:
9353 * gst/rtsp-server/rtsp-session.c:
9354 * gst/rtsp-server/rtsp-stream-transport.c:
9355 * gst/rtsp-server/rtsp-stream.c:
9356 * gst/rtsp-server/rtsp-thread-pool.c:
9357 * gst/rtsp-server/rtsp-token.c:
9360 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9362 * gst/rtsp-server/rtsp-session-pool.c:
9363 * gst/rtsp-server/rtsp-session-pool.h:
9364 session-pool: make vmethod to create a session
9365 Make a vmethod to create a sessions so that subclasses can create
9366 custom session objects
9368 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9370 * gst/rtsp-server/rtsp-auth.c:
9371 * gst/rtsp-server/rtsp-media-factory.h:
9372 * gst/rtsp-server/rtsp-media.h:
9373 * gst/rtsp-server/rtsp-mount-points.h:
9374 * gst/rtsp-server/rtsp-session-pool.h:
9375 * gst/rtsp-server/rtsp-stream.h:
9378 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9380 * docs/libs/gst-rtsp-server-docs.sgml:
9381 * docs/libs/gst-rtsp-server-sections.txt:
9382 * gst/rtsp-server/rtsp-address-pool.c:
9383 * gst/rtsp-server/rtsp-address-pool.h:
9384 * gst/rtsp-server/rtsp-auth.c:
9385 * gst/rtsp-server/rtsp-client.h:
9386 * gst/rtsp-server/rtsp-media-factory.h:
9387 * gst/rtsp-server/rtsp-media.c:
9388 * gst/rtsp-server/rtsp-media.h:
9389 * gst/rtsp-server/rtsp-permissions.c:
9390 * gst/rtsp-server/rtsp-permissions.h:
9391 * gst/rtsp-server/rtsp-server.h:
9392 * gst/rtsp-server/rtsp-session-media.c:
9393 * gst/rtsp-server/rtsp-session-media.h:
9394 * gst/rtsp-server/rtsp-session-pool.h:
9395 * gst/rtsp-server/rtsp-session.h:
9396 * gst/rtsp-server/rtsp-stream-transport.h:
9397 * gst/rtsp-server/rtsp-stream.c:
9398 * gst/rtsp-server/rtsp-thread-pool.h:
9401 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9404 * examples/Makefile.am:
9405 configure: compile cgroup example conditionally
9406 Only compile the cgroup example when we have libcgroup
9408 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * examples/Makefile.am:
9412 * examples/test-cgroups.c:
9413 examples: add cgroups example
9415 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9417 * tests/check/gst/rtspserver.c:
9418 tests: fix compilation
9420 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9422 * gst/rtsp-server/rtsp-thread-pool.c:
9423 thread-pool: fix vmethod invocation
9425 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/rtsp-server/rtsp-thread-pool.c:
9428 * gst/rtsp-server/rtsp-thread-pool.h:
9429 thread-pool: store thread type in thread
9431 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9433 * gst/rtsp-server/rtsp-client.c:
9434 client: pass thread from pool to media _prepare
9435 Get a thread from the configured threadpool and pass it to the prepare method of
9438 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9440 * gst/rtsp-server/rtsp-media.c:
9441 * gst/rtsp-server/rtsp-media.h:
9442 media: Accept a thread in _prepare
9443 Remove out own threadpool handling and use the provided thread and
9444 maincontext for the bus messages and the state changes.
9446 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9448 * gst/rtsp-server/rtsp-server.c:
9449 server: configure client thread pool
9451 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9453 * gst/rtsp-server/rtsp-client.c:
9454 * gst/rtsp-server/rtsp-client.h:
9455 client: add method to configure thread pool
9457 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9459 * gst/rtsp-server/rtsp-client.h:
9460 * gst/rtsp-server/rtsp-server.c:
9461 * gst/rtsp-server/rtsp-server.h:
9462 server: use thread pool
9463 Use the thread pool instead of doing our own thing.
9465 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9467 * gst/rtsp-server/Makefile.am:
9468 * gst/rtsp-server/rtsp-thread-pool.c:
9469 * gst/rtsp-server/rtsp-thread-pool.h:
9470 thread-pool: add object to manage threads
9471 Add an object to manage the client and media threads.
9473 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9475 * gst/rtsp-server/rtsp-auth.c:
9476 auth: debug authorization check
9478 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9480 * gst/rtsp-server/rtsp-media.c:
9481 media: start media pipeline in context
9482 Start the media pipeline in the provided context (or our default one
9483 when NULL). This makes sure that we run the bus thread in this context and that
9484 all media threads are children of this context.
9486 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9488 * gst/rtsp-server/rtsp-media-factory.c:
9489 factory: pass permissions to media by default
9491 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9493 * examples/test-auth.c:
9494 test: add permissions to auth test
9495 Ass some permissions to the media factory in the test.
9497 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9499 * gst/rtsp-server/rtsp-auth.c:
9500 * gst/rtsp-server/rtsp-auth.h:
9501 * gst/rtsp-server/rtsp-client.c:
9502 auth: simplify auth checks
9503 Remove client from methods, it's now in the state
9504 Perform the check specified by the string, use the information from the
9505 thread local context.
9507 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9509 * gst/rtsp-server/rtsp-client.c:
9510 * gst/rtsp-server/rtsp-client.h:
9511 client: add state to current thread
9512 Add the client to the ClientState object.
9513 Place the ClientState on the current thread.
9515 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9517 * gst/rtsp-server/rtsp-media-factory.c:
9518 * gst/rtsp-server/rtsp-media-factory.h:
9519 * gst/rtsp-server/rtsp-media.c:
9520 * gst/rtsp-server/rtsp-media.h:
9521 media: make it possible to set permissions
9522 Make it possible to set permissions on media and media factory objects
9524 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9526 * gst/rtsp-server/Makefile.am:
9527 * gst/rtsp-server/rtsp-permissions.c:
9528 * gst/rtsp-server/rtsp-permissions.h:
9529 permissions: add permissions object
9530 Add a mini object to store permissions based on a role.
9532 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9534 * examples/test-auth.c:
9535 * gst/rtsp-server/rtsp-auth.c:
9536 * gst/rtsp-server/rtsp-auth.h:
9537 * gst/rtsp-server/rtsp-client.c:
9538 auth: add auth checks
9539 Add an enum with auth checks and implement the checks in the auth object.
9540 Perform the checks from the client.
9542 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9544 * examples/test-auth.c:
9545 * gst/rtsp-server/rtsp-auth.c:
9546 * gst/rtsp-server/rtsp-auth.h:
9547 * gst/rtsp-server/rtsp-client.h:
9548 auth: use the token after authentication
9549 After we authenticated a user, keep the Token around in the state.
9551 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9553 * gst/rtsp-server/rtsp-client.c:
9554 * gst/rtsp-server/rtsp-media.c:
9555 * gst/rtsp-server/rtsp-media.h:
9556 * tests/check/gst/media.c:
9557 media: add optional context for bus messages
9558 Add an optional mainloop to _prepare that will handle the bus messages instead
9559 of always using the shared mainloop.
9561 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9563 * gst/rtsp-server/Makefile.am:
9564 * gst/rtsp-server/rtsp-token.c:
9565 * gst/rtsp-server/rtsp-token.h:
9566 token: add authorization token
9567 Add a simply miniobject that contains the authorizations. The object contains a
9568 GstStructure that hold all authorization fields. When a user is authenticated,
9569 the auth module will create a Token for the user. The token is then used to
9570 check what operations the user is allowed to do and various other configuration
9573 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9575 * examples/test-auth.c:
9576 * gst/rtsp-server/rtsp-auth.c:
9577 * gst/rtsp-server/rtsp-auth.h:
9578 * gst/rtsp-server/rtsp-client.c:
9579 * gst/rtsp-server/rtsp-client.h:
9580 * gst/rtsp-server/rtsp-media-factory.c:
9581 * gst/rtsp-server/rtsp-media-factory.h:
9582 * gst/rtsp-server/rtsp-media.c:
9583 * gst/rtsp-server/rtsp-media.h:
9584 auth: remove auth from media and factory
9585 Remove the auth object from media and factory. We want to have the RTSPClient
9586 authenticate and authorize resources, there is no need to place another auth
9587 manager on the media/factory.
9589 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9591 * examples/test-auth.c:
9592 * gst/rtsp-server/rtsp-auth.c:
9593 * gst/rtsp-server/rtsp-auth.h:
9594 * gst/rtsp-server/rtsp-client.h:
9595 auth: add support for multiple basic auth tokens
9596 Make it possible to add multiple basic authorisation tokens to one authorization
9597 object. Associate with each token an authorization group that will define what
9598 capabilities are allowed.
9600 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9602 * gst/rtsp-server/rtsp-client.c:
9603 client: error out on non-aggregate control
9604 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9606 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9608 * gst/rtsp-server/rtsp-client.c:
9609 client: rework setup request a little
9610 Cache the media in DESCRIBE based on the longest matching path with the uri
9611 that we can find in the mount points.
9612 Rework the setup request a little to get the media from the session or from
9613 the longest matching path, this way we can derive the control string as
9614 everything after the path instead of hardcoding it.
9615 Find the stream based on the control string and only open a session when all
9618 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9620 * gst/rtsp-server/rtsp-media.c:
9621 * gst/rtsp-server/rtsp-media.h:
9622 media: add method to find a stream by control url
9624 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9626 * gst/rtsp-server/rtsp-stream.c:
9627 * gst/rtsp-server/rtsp-stream.h:
9628 stream: add method to check control url of stream
9630 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9632 * gst/rtsp-server/rtsp-client.c:
9633 * gst/rtsp-server/rtsp-session-media.c:
9634 * gst/rtsp-server/rtsp-session-media.h:
9635 * gst/rtsp-server/rtsp-session.c:
9636 * gst/rtsp-server/rtsp-session.h:
9637 session: use path matching for session media
9638 Use a path string instead of a uri to lookup session media in the sessions. Also
9639 use path matching to find the largest possible path that matches.
9641 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 * gst/rtsp-server/rtsp-client.c:
9644 * gst/rtsp-server/rtsp-mount-points.c:
9645 * gst/rtsp-server/rtsp-mount-points.h:
9646 * tests/check/gst/mountpoints.c:
9647 mount-points: remove useless vmethod
9648 Making lookups in the mount points should not be done with a URL, if there is a
9649 mapping to be done from URL to mount points, we'll need to do it somewhere
9652 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9654 * gst/rtsp-server/rtsp-mount-points.c:
9655 * gst/rtsp-server/rtsp-mount-points.h:
9656 * tests/check/gst/mountpoints.c:
9657 mount-points: improve mount point searching
9658 Use a GSequence to keep track of the mount points.
9659 Match a URL to the longest matching registered mount point. This should be the
9660 URL to perform aggreagate control and the remainder is the stream specific
9662 Add some unit tests for this.
9664 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9666 * gst/rtsp-server/Makefile.am:
9667 rtsp-server: Allow building of static library
9669 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * tests/check/gst/mediafactory.c:
9672 tests: fix compilation
9674 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9676 * gst/rtsp-server/rtsp-sdp.c:
9677 sdp: get control string from stream
9678 Use the control string as configured in the stream.
9680 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9682 * gst/rtsp-server/rtsp-stream.c:
9683 * gst/rtsp-server/rtsp-stream.h:
9684 stream: add methods and property to set control string
9686 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9688 * gst/rtsp-server/rtsp-client.c:
9690 Rename variables for clarity
9691 Keep media in state when we can
9693 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9695 * gst/rtsp-server/rtsp-client.c:
9696 * gst/rtsp-server/rtsp-stream.c:
9697 * gst/rtsp-server/rtsp-stream.h:
9698 stream: add more support for IPv6
9699 Rename _get_address to _get_multicast_address in GstRTSPStream to
9700 make it clear that this function only deals with multicast.
9701 Make it possible to have both an IPv4 and IPv6 multicast address on
9702 a stream. Give the client an IPv4 or IPv6 address depending on the
9703 address it used to connect to the server.
9704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9706 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9708 * gst/rtsp-server/rtsp-client.c:
9711 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9713 * gst/rtsp-server/rtsp-stream.c:
9714 stream: handle failed port allocation
9715 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9716 can't allocate any family at all. Also keep track of what port families we
9718 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9720 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9722 * gst/rtsp-server/rtsp-stream.c:
9723 stream: improve docs
9725 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9727 * gst/rtsp-server/rtsp-stream-transport.c:
9728 stream-transport: remove old if 0 block
9730 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9732 * tests/check/gst/client.c:
9734 gst_rtsp_client_get_uri() has been removed
9735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9737 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9739 * gst/rtsp-server/rtsp-client.c:
9740 * gst/rtsp-server/rtsp-client.h:
9741 client: add method to filter managed sessions
9742 Add a method to filter the sessions managed by this client connection.
9743 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9745 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9747 * gst/rtsp-server/rtsp-client.c:
9748 * gst/rtsp-server/rtsp-client.h:
9749 client: remove _get_uri() method
9750 Remove the get_uri() method on the client. A client has no uri, the uri
9751 property is an internal property to manage the last cached media for
9754 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9756 * gst/rtsp-server/rtsp-media-factory.h:
9757 media-factory: fix typo
9759 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9761 * gst/rtsp-server/rtsp-media.c:
9762 rtsp-media: Do not leak the query in default_query_stop
9763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9765 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9767 * gst/rtsp-server/rtsp-media.c:
9768 media: don't unlock when conversion fails
9769 Don't unlock the state lock when conversion fails because it was not locked.
9771 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9773 * gst/rtsp-server/rtsp-media.c:
9774 * gst/rtsp-server/rtsp-media.h:
9775 Add query_position and query_stop vmethods to rtsp-media
9777 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9779 * gst/rtsp-server/rtsp-media.c:
9780 Fix typo in property install for rtsp-media's time-provider
9782 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9784 * gst/rtsp-server/rtsp-client.c:
9785 * gst/rtsp-server/rtsp-client.h:
9786 client: clean some variables
9787 Clean some variables and add some guards to _send_request()
9789 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9791 * gst/rtsp-server/rtsp-client.c:
9792 * gst/rtsp-server/rtsp-client.h:
9793 Add gst_rtsp_client_send_request API
9794 This makes it possible to send arbitrary messages to a client, such as
9795 SET_PARAMETER or GET_PARAMETER
9797 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9799 * gst/rtsp-server/rtsp-media.c:
9800 * gst/rtsp-server/rtsp-media.h:
9801 media: add _get_element() method
9802 Add method to get the element used when creating the media.
9803 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9805 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9807 * gst/rtsp-server/rtsp-media.c:
9810 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9812 * gst/rtsp-server/rtsp-stream.c:
9813 * gst/rtsp-server/rtsp-stream.h:
9814 stream: allow access to the rtp session
9815 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9817 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9819 * gst/rtsp-server/rtsp-stream.c:
9820 * gst/rtsp-server/rtsp-stream.h:
9821 dscp qos support in gst-rtsp-stream
9822 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9824 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9826 * tests/check/gst/rtspserver.c:
9828 Actually do what the comment says. Also keep the old code around, not sure what
9829 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9830 it currently doesn't.
9832 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9834 * gst/rtsp-server/rtsp-client.c:
9835 client: also watch newly created session
9836 When we newly created a session, start watching it immediately instead of
9837 on the next request.
9839 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9841 * tests/check/gst/client.c:
9842 tests: add unit test for new-session
9843 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9845 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9847 * gst/rtsp-server/rtsp-client.c:
9848 client: emit new-session when new session is created
9849 Only emit new-session when we created a new session for a client, not when a
9850 client picked up a previous session.
9851 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9853 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9855 * gst/rtsp-server/rtsp-client.c:
9856 client: handle asterisk as path in requests
9857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9859 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9861 * gst/rtsp-server/rtsp-media.c:
9862 media: handle segment query format mismatch
9863 It's possible that the segment query returns with a different format than what
9864 we asked for, handle this case also.
9866 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9868 * gst/rtsp-server/rtsp-media.c:
9869 media: use segment stop in collect_media_stats
9870 Use segment stop instead of duration as range end point.
9871 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9873 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9875 * gst/rtsp-server/rtsp-media.c:
9876 * tests/check/gst/media.c:
9877 rtsp-media: Do not leak the element in take_pipeline
9878 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9880 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9882 * gst/rtsp-server/rtsp-client.c:
9883 * gst/rtsp-server/rtsp-client.h:
9884 rtsp-client: Make configure_client_transport virtual
9885 This patch makes configure_client_transport virtual. The functionality is
9886 needed to handle some weird clients sending multicast transport settings as url
9888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9890 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9892 * gst/rtsp-server/rtsp-client.c:
9893 * gst/rtsp-server/rtsp-client.h:
9894 rtsp-client: Make param_set and param_get virtual
9895 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9897 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9899 * gst/rtsp-server/rtsp-client.c:
9900 * gst/rtsp-server/rtsp-media.c:
9901 * gst/rtsp-server/rtsp-media.h:
9902 media: convert_range replaces get_range_times
9903 get_range_times worked for handling UTC ranges for seeks, but we also
9904 need to convert back from NPT to the requested unit in
9905 get_range_string. convert_range is now used for both.
9906 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9908 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9910 * gst/rtsp-server/rtsp-client.c:
9911 * gst/rtsp-server/rtsp-sdp.c:
9912 * gst/rtsp-server/rtsp-sdp.h:
9913 sdp: cleanup sdp info
9914 We don't need to pass the proto, we can more easily check a boolean.
9915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9917 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9919 * gst/rtsp-server/rtsp-sdp.c:
9920 use 0.0.0.0 or :: for c= line instead of server address
9922 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9924 * gst/rtsp-server/rtsp-client.c:
9925 use local address, not remote, in SDP
9926 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9928 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9931 Automatic update of common submodule
9932 From 098c0d7 to 01a7a46
9934 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9936 * gst/rtsp-server/rtsp-media.c:
9937 * gst/rtsp-server/rtsp-media.h:
9938 media: possibility to override range time conversion
9939 Make it possible to override the conversion from GstRTSPTimeRange to
9940 GstClockTimes, that is done before seeking on the media
9941 pipeline. Overriding can be useful for UTC ranges, where the default
9942 conversion gives nanoseconds since 1900.
9943 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9945 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9947 * gst/rtsp-server/rtsp-server.c:
9948 * gst/rtsp-server/rtsp-server.h:
9949 rtsp-server: Expose the use_client_settings API
9950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9952 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9954 * gst/rtsp-server/rtsp-client.c:
9955 * gst/rtsp-server/rtsp-stream.c:
9956 * gst/rtsp-server/rtsp-stream.h:
9957 rtspstream: handle both ipv4 and ipv6 clients
9958 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9960 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9962 * gst/rtsp-server/rtsp-sdp.c:
9963 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9964 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9965 We already have a way to place extra attributes in the SDP by using a string
9966 property with prefix x- or a- in the caps.
9968 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9970 * gst/rtsp-server/rtsp-sdp.c:
9971 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9972 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9973 We already have a way to place extra attributes in the SDP, just make a string
9974 property in the payloader with a- or x- prefix.
9976 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9978 * gst/rtsp-server/rtsp-sdp.c:
9979 rtsp: place a- and x- properties as attributes
9980 application/x-rtp has properties with a- and x- prefixes that should be
9981 placed as attributes in the SDP for the media instead of being added to the
9984 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9986 * examples/Makefile.am:
9987 * examples/test-video.c:
9988 example: add TLS example
9990 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9992 * gst/rtsp-server/rtsp-server.c:
9993 * gst/rtsp-server/rtsp-server.h:
9994 server: add support for TLS
9995 Add methods to set and get a TLS certificate.
9996 Add vmethod to configure a new connection. By default, configure the TLS
9997 certificate in a new connection if needed.
9999 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10001 * gst/rtsp-server/rtsp-server.c:
10002 * gst/rtsp-server/rtsp-server.h:
10003 server: remove accept_client vmethod
10004 This vmethod is not very useful so remove it.
10006 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10008 * gst/rtsp-server/rtsp-server.c:
10009 server: don't crash on NULL GError
10011 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
10013 * gst/rtsp-server/rtsp-session-pool.c:
10014 rtsp-session-pool: corrected session timeout detection
10015 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
10017 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10019 * gst/rtsp-server/rtsp-client.c:
10020 client: improve debug
10022 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10024 * gst/rtsp-server/rtsp-client.c:
10025 * gst/rtsp-server/rtsp-client.h:
10026 * gst/rtsp-server/rtsp-server.c:
10027 server: refactor connection setup
10028 Let the server accept the socket connection and construct a GstRTSPConnection
10029 from it. Remove the code from the client and let the client only deal with
10030 a fully configure GstRTSPConnection object.
10031 We will need this later when the server will configure the connection for
10034 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10036 * gst/rtsp-server/rtsp-stream.c:
10037 stream: keep the transport object alive
10038 Keep the transport object alive while we have it as qdata on the
10041 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
10043 * gst/rtsp-server/rtsp-client.c:
10044 * gst/rtsp-server/rtsp-server.c:
10045 rtsp-server: Do not crash on nmapping of server
10046 * generate error when gst_rtsp_connection_accept fails
10047 * do not stop accepting incoming connections because
10048 accepting a client fails
10049 https://bugzilla.gnome.org/show_bug.cgi?id=701072
10051 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
10053 * gst/rtsp-server/rtsp-client.c:
10054 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
10055 https://bugzilla.gnome.org/show_bug.cgi?id=700953
10057 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
10059 * gst/rtsp-server/rtsp-sdp.c:
10060 rtsp-sdp: Parse framerate caps field and set SDP attribute
10061 The SDP attribute and its format is described in RFC4566.
10062 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10064 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
10066 * gst/rtsp-server/rtsp-sdp.c:
10067 rtsp-sdp: Parse width/height from caps and set SDP attribute
10068 The SDP attribute and its format is described in RFC6064.
10069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10071 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
10073 * gst/rtsp-server/rtsp-sdp.c:
10074 * tests/check/gst/client.c:
10075 rtsp-sdp: add bandwidth line
10076 https://bugzilla.gnome.org/show_bug.cgi?id=699220
10078 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10081 Automatic update of common submodule
10082 From 5edcd85 to 098c0d7
10084 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10086 * tests/check/gst/media.c:
10087 tests: add dynamic payloader prepare/unprepare check
10089 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10091 * gst/rtsp-server/rtsp-media.c:
10092 media: release lock when removing fakesink
10094 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10096 * gst/rtsp-server/rtsp-stream.c:
10097 stream: set elements to NULL before removing
10098 When removing a stream, set the elements to NULL first. This avoids
10099 element-is-not-in-NULL-state errors when we dispose the elements.
10101 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
10104 Automatic update of common submodule
10105 From 3cb3d3c to 5edcd85
10107 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10109 * gst/rtsp-server/rtsp-media.c:
10110 * gst/rtsp-server/rtsp-media.h:
10111 media: listen to pad-removed signals
10112 Listen to the pad-removed signal and remove the stream associated with the
10114 Add signal to be notified of the removed pad.
10115 Remove the fakesink in unprepare()
10116 Fix signatures of the signal methods
10118 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10120 * examples/test-sdp.c:
10121 tests: add example of reusable pipelines
10123 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
10125 * gst/rtsp-server/rtsp-stream.c:
10126 * gst/rtsp-server/rtsp-stream.h:
10127 stream: add method to get the srcpad
10129 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10131 * tests/check/gst/media.c:
10132 check: add media prepare/unprepare test
10133 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10135 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
10137 * gst/rtsp-server/rtsp-media.c:
10138 media: disconnect from signal handlers in unprepare()
10139 We connected to the pad-added and no-more-pads signals in prepare() so
10140 we need to disconnect from them in unprepare().
10141 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10143 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
10145 * gst/rtsp-server/rtsp-media.c:
10146 media: don't free streams array
10147 Don't free the streams array in the unprepare() method, they were not
10148 added in prepare().
10149 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10151 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
10153 * gst/rtsp-server/rtsp-media.c:
10154 media: don't unref the pipeline in unprepare
10155 Unprepare() should undo what prepare() does. Because the pipeline is
10156 not created in prepare(), we should not unref it in unprepare()
10158 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
10160 * gst/rtsp-server/rtsp-stream.c:
10161 stream: clear session and caps for reuse
10162 Set the session and caps to NULL after unref otherwise we might unref
10164 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10166 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
10168 * gst/rtsp-server/rtsp-client.c:
10169 client: send out teardown signal before tearing down
10170 The advantage is that in the signal handler you get direct access to
10171 information about what streams are about to get torn down (in the
10172 GstRTSPClientState).
10173 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
10175 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
10177 * gst/rtsp-server/rtsp-client.c:
10178 * gst/rtsp-server/rtsp-client.h:
10179 client: expose connection
10180 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
10182 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
10185 Automatic update of common submodule
10186 From aed87ae to 3cb3d3c
10188 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10190 * gst/rtsp-server/rtsp-media.c:
10191 * gst/rtsp-server/rtsp-media.h:
10192 * gst/rtsp-server/rtsp-session-media.c:
10193 * gst/rtsp-server/rtsp-session-media.h:
10194 media: add method to get the base_time of the pipeline
10195 Together with a shared clock, this base-time could eventually be sent to
10196 the client so that it can reconstruct the exact running-time of the clock
10199 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10201 * gst/rtsp-server/Makefile.am:
10202 * gst/rtsp-server/rtsp-media.c:
10203 * gst/rtsp-server/rtsp-media.h:
10204 * gst/rtsp-server/rtsp-sdp.c:
10205 media: add GstNetTimeProvider support
10206 Add a property to let the media provide a GstNetTimeProvider for its clock.
10207 Make methods to get the clock and nettimeprovider
10208 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
10209 provider and also the current time of the clock. This should make it possible
10210 for (GStreamer) clients to slave their clock to the server clock.
10212 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
10215 Automatic update of common submodule
10216 From 04c7a1e to aed87ae
10218 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10220 * gst/rtsp-server/rtsp-media.c:
10221 media: wait for buffering to complete
10222 Wait for buffering to complete before changing the state to the target state.
10224 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10226 * gst/rtsp-server/rtsp-media.c:
10227 media: small cleanup
10229 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
10231 * tests/check/gst/rtspserver.c:
10232 tests: remove extra unref in test_setup_non_existing_stream
10233 The unref is not needed anymore, teardown runs without it.
10234 https://bugzilla.gnome.org/show_bug.cgi?id=696542
10236 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
10238 * tests/check/gst/rtspserver.c:
10239 tests: GSocketService cleanup in test_bind_already_in_use
10240 Use g_socket_service_stop so the rtspserver test stops listening for
10241 incoming connections in test_bind_already_in_use.
10242 https://bugzilla.gnome.org/show_bug.cgi?id=696541
10244 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
10246 * gst/rtsp-server/rtsp-media-factory.c:
10247 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
10248 Instead use a GWeakRef which is safe to use
10249 This is a known GLib bug, see:
10250 https://bugzilla.gnome.org/show_bug.cgi?id=667145
10252 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
10254 * gst/rtsp-server/rtsp-client.c:
10255 * gst/rtsp-server/rtsp-media.c:
10256 * gst/rtsp-server/rtsp-media.h:
10257 * gst/rtsp-server/rtsp-sdp.c:
10258 * tests/check/gst/media.c:
10259 * tests/check/gst/rtspserver.c:
10260 rtsp-media/client: Reply to PLAY request with same type of Range
10261 Remember the type of Range from the PLAY request and use the same type for
10264 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
10266 * gst/rtsp-server/rtsp-client.c:
10267 * gst/rtsp-server/rtsp-client.h:
10268 * tests/check/gst/client.c:
10269 rtsp-client: expose uri
10271 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
10273 * tests/check/gst/mediafactory.c:
10274 tests: Hold ref while creating second media
10275 To test if the media aren't shared, make sure we keep the first one while creating a second
10276 otherwise the same memory address may be reused.
10278 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
10281 configure: remove out-of-date comment
10283 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10286 .gitignore: ignore more build files
10288 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10290 * tests/check/Makefile.am:
10291 tests: use right _LIBS variable for gst-plugins-base libs
10293 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10295 * tests/check/Makefile.am:
10296 check: add librtp to libs
10298 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10300 * tests/check/gst/rtspserver.c:
10301 tests: Add test to check selecting a port the server will send from
10303 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10305 * tests/check/gst/rtspserver.c:
10306 tests: Make sure packets are actually received
10308 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10310 * gst/rtsp-server/rtsp-stream.c:
10311 stream: Select unicast address from pool if appropriate
10313 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10315 * gst/rtsp-server/rtsp-stream.c:
10316 stream: Properties are always there in Gst 1.0
10318 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10320 * tests/check/gst/addresspool.c:
10321 tests: Add tests for unicast addresses in pool
10323 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10325 * gst/rtsp-server/rtsp-address-pool.c:
10326 * tests/check/gst/addresspool.c:
10327 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10329 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10331 * docs/libs/gst-rtsp-server-sections.txt:
10332 * gst/rtsp-server/rtsp-address-pool.c:
10333 * gst/rtsp-server/rtsp-address-pool.h:
10334 * gst/rtsp-server/rtsp-stream.c:
10335 * tests/check/gst/addresspool.c:
10336 address-pool: Add unicast addresses
10338 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10341 * gst/rtsp-server/rtsp-server.c:
10342 * tests/check/gst/rtspserver.c:
10343 rtsp-server: Limit the number of threads per server instance
10344 If we exceed the maximum, just round robin the clients over the existing
10347 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10349 * gst/rtsp-server/rtsp-server.c:
10350 rtsp-server: No need to store the GMainContext in the client context
10352 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10354 * tests/check/gst/rtspserver.c:
10355 tests: Add test for client disconnection
10357 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10359 * tests/check/gst/rtspserver.c:
10360 tests: Test client and session timeouts with multiple threads
10362 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10364 * gst/rtsp-server/rtsp-address-pool.c:
10365 * gst/rtsp-server/rtsp-auth.c:
10366 * gst/rtsp-server/rtsp-client.c:
10367 * gst/rtsp-server/rtsp-media-factory-uri.c:
10368 * gst/rtsp-server/rtsp-media-factory.c:
10369 * gst/rtsp-server/rtsp-media.c:
10370 * gst/rtsp-server/rtsp-mount-points.c:
10371 * gst/rtsp-server/rtsp-server.c:
10372 * gst/rtsp-server/rtsp-session-media.c:
10373 * gst/rtsp-server/rtsp-session-pool.c:
10374 * gst/rtsp-server/rtsp-session.c:
10375 Document locking and its order
10377 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10379 * tests/check/gst/rtspserver.c:
10380 tests: Test that slow DESCRIBE don't block other clients
10382 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10384 * tests/check/gst/client.c:
10385 tests: Add tests for client-requested multicast address
10387 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10389 * docs/libs/gst-rtsp-server-sections.txt:
10390 docs: Put the various functions in the right sections
10392 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10394 * docs/libs/gst-rtsp-server-docs.sgml:
10395 * docs/libs/gst-rtsp-server-sections.txt:
10396 * gst/rtsp-server/rtsp-address-pool.c:
10397 * gst/rtsp-server/rtsp-address-pool.h:
10398 docs: Generate docs for GstRTSPAddressPool
10400 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10402 * gst/rtsp-server/rtsp-client.c:
10403 * gst/rtsp-server/rtsp-stream.c:
10404 * gst/rtsp-server/rtsp-stream.h:
10405 client: Check client provided addresses against the address pool
10407 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10409 * gst/rtsp-server/rtsp-address-pool.c:
10410 * gst/rtsp-server/rtsp-address-pool.h:
10411 * tests/check/gst/addresspool.c:
10412 address-pool: Add API to request a specific address from the pool
10413 Also add relevant unit tests.
10415 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10417 * tests/check/gst/mediafactory.c:
10418 tests: Check the passing around of a RTSPAddressPool
10419 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10420 way down to the stream.
10422 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10424 * tests/check/gst/addresspool.c:
10425 tests: Add more tests for the address pool
10427 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10429 * gst/rtsp-server/rtsp-address-pool.c:
10430 address-pool: Fix off by one error
10431 When splitting a port range, the port after a skip is not part of range.
10433 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10436 Automatic update of common submodule
10437 From 2de221c to 04c7a1e
10439 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10442 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10443 AM_CONFIG_HEADER was removed in automake 1.13
10444 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10446 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10449 Automatic update of common submodule
10450 From a942293 to 2de221c
10452 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10454 * gst/rtsp-server/rtsp-client.c:
10455 client: make sure the watch exists while sending data
10456 Protect the send_func with a lock. This allows us to wait for sending
10457 to complete before changing the send_func and user_data. We add an
10458 extra ref to the watch to make sure that it remains valid during
10460 When closing the connection, set the send_func to NULL
10461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10463 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10465 * tests/check/Makefile.am:
10466 tests: use GST_*_1_0 environment variables everywhere
10467 The _1_0 suffixed environment variables override the
10468 non-suffixed ones, so if we're in an environment that
10469 sets the _1_0 suffixed ones, such as jhbuild, we need
10470 to set those to make sure ours actually always get
10473 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10476 Automatic update of common submodule
10477 From acb04d9 to a942293
10479 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10481 * gst/rtsp-server/rtsp-client.c:
10482 rtsp-client: set the client backlog
10483 Set the client backlog to a reasonable default
10485 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10487 * gst/rtsp-server/rtsp-media.c:
10488 rtsp-media: Make the element a constructor parameter
10489 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10491 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10493 * docs/libs/Makefile.am:
10494 docs: Link with gcov library when gcov is enabled
10495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10497 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10499 * gst/rtsp-server/rtsp-media.c:
10500 media: match prepare with unprepare
10501 Really unprepare when there were an equal amount of prepare calls.
10503 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10505 * gst/rtsp-server/rtsp-media.c:
10506 media: media has to be unprepared in finalize
10507 Because unprepare takes away the last ref on the media.
10509 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10511 * gst/rtsp-server/rtsp-client.c:
10512 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10513 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10514 We can't use the refcount to trigger unprepare because it is the unprepare call
10515 that removes the last refcount after all messages are consumed. What we should
10516 probably do is make a prepared refcount and only unprepare when the refcount
10519 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10521 * gst/rtsp-server/rtsp-media.c:
10522 media: let the source unref the last media ref
10523 the last ref to the media is held by the source so we don't need to add more ref
10524 and unrefs, we simply destroy the media when the source is gone.
10526 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10528 * gst/rtsp-server/rtsp-media.c:
10529 media: improve debug
10531 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10533 * gst/rtsp-server/rtsp-media.c:
10535 Make sure we are in the right state when collecting the position and duration.
10536 Only make ourselves PREPARED when we were previously PREPARING.
10538 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10540 * gst/rtsp-server/rtsp-media.c:
10541 media: use g_object_ref/unref for GObjects
10543 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10545 * gst/rtsp-server/rtsp-client.c:
10546 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10547 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10548 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10549 isn't being used anymore.
10551 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10553 * gst/rtsp-server/rtsp-media.c:
10554 Fix compiler warning
10556 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10558 * gst/rtsp-server/rtsp-media-factory-uri.c:
10559 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10561 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10563 * gst/rtsp-server/rtsp-session-media.h:
10566 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10568 * gst/rtsp-server/rtsp-media.c:
10569 * tests/check/gst/media.c:
10570 media: avoid element leak
10572 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10574 * gst/rtsp-server/rtsp-media.c:
10575 media: require an element in media constructor
10577 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10579 * gst/rtsp-server/rtsp-client.c:
10580 Revert "client: TEARDOWN brings that state to Init again"
10581 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10582 The object is already disposed, there is no point in setting the state.
10584 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10586 * gst/rtsp-server/rtsp-client.c:
10587 client: TEARDOWN brings that state to Init again
10589 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10591 * docs/libs/gst-rtsp-server-sections.txt:
10592 * examples/test-auth.c:
10593 * gst/rtsp-server/rtsp-auth.c:
10594 * gst/rtsp-server/rtsp-auth.h:
10595 * gst/rtsp-server/rtsp-client.c:
10596 * gst/rtsp-server/rtsp-client.h:
10597 * gst/rtsp-server/rtsp-media-factory-uri.c:
10598 * gst/rtsp-server/rtsp-media-factory-uri.h:
10599 * gst/rtsp-server/rtsp-media-factory.c:
10600 * gst/rtsp-server/rtsp-media-factory.h:
10601 * gst/rtsp-server/rtsp-media.c:
10602 * gst/rtsp-server/rtsp-media.h:
10603 * gst/rtsp-server/rtsp-mount-points.c:
10604 * gst/rtsp-server/rtsp-mount-points.h:
10605 * gst/rtsp-server/rtsp-sdp.c:
10606 * gst/rtsp-server/rtsp-server.c:
10607 * gst/rtsp-server/rtsp-server.h:
10608 * gst/rtsp-server/rtsp-session-media.c:
10609 * gst/rtsp-server/rtsp-session-media.h:
10610 * gst/rtsp-server/rtsp-session-pool.c:
10611 * gst/rtsp-server/rtsp-session-pool.h:
10612 * gst/rtsp-server/rtsp-session.c:
10613 * gst/rtsp-server/rtsp-session.h:
10614 * gst/rtsp-server/rtsp-stream-transport.c:
10615 * gst/rtsp-server/rtsp-stream-transport.h:
10616 * gst/rtsp-server/rtsp-stream.c:
10617 * gst/rtsp-server/rtsp-stream.h:
10618 * tests/check/gst/media.c:
10619 rtsp: make object details private
10620 Make all object details private
10621 Add methods to access private bits
10623 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10625 * tests/check/Makefile.am:
10626 * tests/check/gst/media.c:
10627 tests: add media tests
10629 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10631 * gst/rtsp-server/rtsp-media.c:
10632 media: check if prepared for some methods
10633 Check that the media object is prepared before doing seek and getting the
10634 current position etc.
10635 Add some g_return checks.
10637 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10639 * tests/check/Makefile.am:
10640 * tests/check/gst/mediafactory.c:
10641 tests: add mediafactory test
10643 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10645 * gst/rtsp-server/rtsp-stream.c:
10646 stream: improve debug
10648 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10650 * gst/rtsp-server/rtsp-media.c:
10651 * gst/rtsp-server/rtsp-media.h:
10652 media: unref pipeline in finalize to avoid leaking it
10654 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10656 * gst/rtsp-server/rtsp-media-factory-uri.c:
10657 * gst/rtsp-server/rtsp-media.c:
10658 rtsp: use gst_object_unref on GstObjects
10660 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10662 * gst/rtsp-server/rtsp-media-factory.c:
10663 media-factory: require an url
10665 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10667 * examples/test-uri.c:
10668 examples: fix include
10670 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10672 * gst/rtsp-server/rtsp-server.h:
10673 server: remove unused include
10675 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10677 * tests/check/Makefile.am:
10678 * tests/check/gst/mountpoints.c:
10679 tests: add test for mountpoints
10681 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10683 * gst/rtsp-server/rtsp-client.c:
10684 client: fix factory leak
10685 Keep the factory in the state object only for authorization checks and make
10686 sure we unref it on failure. Also don't keep invalid objects in the state
10689 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10691 * gst/rtsp-server/rtsp-mount-points.c:
10692 mounts: add g_return_if guards
10694 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10696 * tests/check/gst/client.c:
10697 tests: add more tests
10699 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10701 * gst/rtsp-server/rtsp-client.c:
10702 client: improve debug
10704 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10706 * gst/rtsp-server/rtsp-client.c:
10707 client: improve debug and fix leaks
10708 Cleanup the uri and session when there is a bad request.
10710 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10715 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10717 * tests/check/gst/client.c:
10718 test: add test for session in options request
10720 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10722 * gst/rtsp-server/rtsp-client.c:
10723 client: use 454 when session can't be found
10724 We should use 454 when a session can't be found because there was no session
10725 pool configured in the server. This is not a server configuration problem
10726 because the server on which the request is done might not be the same one that
10727 will keep the sessions for us and so it does not need to support sessions.
10729 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10731 * gst/rtsp-server/rtsp-client.c:
10732 client: only free connection when there is one
10733 It's possible that the client doesn't have a connection when we try to free it.
10735 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10737 * tests/check/Makefile.am:
10738 * tests/check/gst/client.c:
10739 tests: add unit test for the client object
10741 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10743 * gst/rtsp-server/rtsp-client.c:
10744 client: small cleanup
10746 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10748 * gst/rtsp-server/rtsp-client.h:
10749 client: remove unused include
10751 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10753 * gst/rtsp-server/rtsp-client.c:
10754 client: fix compilation
10756 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10758 * gst/rtsp-server/rtsp-client.c:
10759 client: call destroy without the lock
10761 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10763 * gst/rtsp-server/rtsp-client.c:
10764 * gst/rtsp-server/rtsp-client.h:
10765 client: make the client usable without a socket
10766 Make a method to let the client handle a message and a callback when the client
10767 wants us to send a response message back. This makes it possible to also use the
10768 client object without the sockets, which should make it easier to test.
10770 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10772 * gst/rtsp-server/rtsp-client.c:
10773 * gst/rtsp-server/rtsp-client.h:
10774 client: small cleanup
10776 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10778 * docs/libs/gst-rtsp-server-sections.txt:
10779 * gst/rtsp-server/rtsp-client.c:
10780 * gst/rtsp-server/rtsp-client.h:
10781 * gst/rtsp-server/rtsp-server.c:
10782 client: remove reference to server
10783 We don't need to keep a ref to the server
10785 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10787 * gst/rtsp-server/rtsp-client.c:
10788 * gst/rtsp-server/rtsp-client.h:
10789 client: add locking
10790 Also add some g_return_if()
10792 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10794 * gst/rtsp-server/rtsp-client.c:
10795 client: log more errors
10797 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10799 * gst/rtsp-server/rtsp-client.c:
10800 client: fix compilation
10802 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10804 * gst/rtsp-server/rtsp-client.c:
10805 * gst/rtsp-server/rtsp-client.h:
10806 client: add generic close-after-send support
10807 Add a property to send_response() to close the connection after the response has
10808 been sent to the client.
10810 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10813 * docs/libs/gst-rtsp-server-docs.sgml:
10814 * docs/libs/gst-rtsp-server-sections.txt:
10815 * docs/libs/gst-rtsp-server.types:
10816 * examples/test-auth.c:
10817 * examples/test-launch.c:
10818 * examples/test-mp4.c:
10819 * examples/test-multicast.c:
10820 * examples/test-multicast2.c:
10821 * examples/test-ogg.c:
10822 * examples/test-readme.c:
10823 * examples/test-sdp.c:
10824 * examples/test-uri.c:
10825 * examples/test-video.c:
10826 * gst/rtsp-server/Makefile.am:
10827 * gst/rtsp-server/rtsp-auth.h:
10828 * gst/rtsp-server/rtsp-client.c:
10829 * gst/rtsp-server/rtsp-client.h:
10830 * gst/rtsp-server/rtsp-media-mapping.c:
10831 * gst/rtsp-server/rtsp-media-mapping.h:
10832 * gst/rtsp-server/rtsp-mount-points.c:
10833 * gst/rtsp-server/rtsp-mount-points.h:
10834 * gst/rtsp-server/rtsp-server.c:
10835 * gst/rtsp-server/rtsp-server.h:
10836 * gst/rtsp-server/rtsp-session-media.c:
10837 * gst/rtsp-server/rtsp-session-pool.c:
10838 * gst/rtsp-server/rtsp-session-pool.h:
10839 * tests/check/gst/rtspserver.c:
10840 MediaMapping -> MountPoints
10841 Describes better what the object manages.
10843 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10846 configure: bump required version of -base
10848 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10850 * gst/rtsp-server/rtsp-media.c:
10853 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10855 * gst/rtsp-server/rtsp-media.c:
10856 * gst/rtsp-server/rtsp-media.h:
10857 media: support more Range formats
10858 Use the new -base methods to convert the Range string into a seek start and stop
10861 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10863 * examples/test-launch.c:
10864 examples: fix whitespace
10866 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10868 * examples/test-auth.c:
10869 test-auth: add example of how to remove sessions
10870 Add an example of the session filter api.
10872 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10874 * examples/test-uri.c:
10875 test-uri: remove mapping example
10877 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10879 * examples/test-uri.c:
10880 test-uri: fix callback signature
10882 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10884 * gst/rtsp-server/rtsp-media-factory.c:
10885 factory: keep ref to factory while media active
10886 While the media from a factory is alive, keep a ref to the factory.
10887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10889 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10891 * gst/rtsp-server/rtsp-media-factory-uri.c:
10892 factory-uri: add some debug
10894 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10896 * gst/rtsp-server/rtsp-stream.c:
10897 stream: set udp sources to PLAYING
10898 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10899 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10901 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10903 * gst/rtsp-server/rtsp-media-factory-uri.c:
10904 factory-uri: take ref to factory
10905 Take a ref to the factory that we place in our list.
10907 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10909 * tests/Makefile.am:
10910 * tests/test-reuse.c:
10911 test: add test for server reuse
10912 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10914 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10916 * gst/rtsp-server/rtsp-server.c:
10917 server: start and stop multiple times
10918 Stop listening on the RTSP port when the GSource is removed, so clients
10919 can't connect and the server can be started again.
10920 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10922 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10924 * gst/rtsp-server/rtsp-server.c:
10925 server: fix small leak
10927 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10929 * gst/rtsp-server/rtsp-media.c:
10930 media: unref source in finish_unprepare
10931 The source is created in prepare, unref it in finish_unprepare.
10932 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10934 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10936 * gst/rtsp-server/rtsp-client.c:
10937 * gst/rtsp-server/rtsp-media.c:
10938 rtsp-media: remove bus watch before finalizing
10939 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10940 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10941 the GDestroyNotify function.
10942 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10943 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10944 gst_rtsp_media_unprepare before unreffing the media.
10945 This way, the bus watch will be removed before the media is finalized.
10946 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10948 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10950 * gst/rtsp-server/rtsp-client.c:
10951 * gst/rtsp-server/rtsp-client.h:
10952 client: wait until the TEARDOWN response is sent to close the connection
10953 Responses can be sent async so we need to wait until the TEARDOWN response has
10954 been written before we close the connection to the client. This avoids the risk
10955 of writing/polling closed sockets.
10956 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10958 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10960 * gst/rtsp-server/rtsp-stream.c:
10961 rtsp-stream: plug socket leak
10962 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10964 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10967 Automatic update of common submodule
10968 From 6bb6951 to a72faea
10970 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10972 * gst/rtsp-server/rtsp-media-factory-uri.c:
10973 rtsp-server: don't use deprecated API
10975 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10977 * gst/rtsp-server/rtsp-client.c:
10978 rtsp-client: fix unused-but-set-variable compiler warning
10979 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10981 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10984 * docs/libs/gst-rtsp-server-sections.txt:
10985 * gst/rtsp-server/rtsp-client.c:
10988 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10990 * examples/Makefile.am:
10991 * examples/test-multicast2.c:
10992 examples: add another multicast example
10993 Add an example for how to configure separate multicast ranges for each media
10996 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10998 * examples/test-multicast.c:
11001 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11003 * gst/rtsp-server/rtsp-client.c:
11004 * gst/rtsp-server/rtsp-media.c:
11005 * gst/rtsp-server/rtsp-session-media.c:
11006 * gst/rtsp-server/rtsp-session-media.h:
11007 * gst/rtsp-server/rtsp-stream-transport.c:
11008 * gst/rtsp-server/rtsp-stream-transport.h:
11009 stream: use the address managed by the stream
11010 Use the address managed by the stream for multicast. This allows us to have 1
11011 multicast address for each stream.
11012 Because the address is now managed by the stream we don't have to pass it around
11014 Set the address pool on the streams.
11016 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11018 * gst/rtsp-server/rtsp-client.c:
11019 * gst/rtsp-server/rtsp-media.c:
11020 * gst/rtsp-server/rtsp-stream.c:
11021 rtsp: improve debug
11023 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11025 * gst/rtsp-server/rtsp-media.c:
11026 * gst/rtsp-server/rtsp-media.h:
11027 media: add signal for new streams
11028 This allows applications to listen for new streams and configure properties on
11029 them, like the address pool.
11031 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11033 * gst/rtsp-server/rtsp-media.c:
11034 media: configure address pool in new streams
11036 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11038 * gst/rtsp-server/rtsp-stream.c:
11039 * gst/rtsp-server/rtsp-stream.h:
11040 stream: add methods to deal with address pool
11041 Add methods to get and set the address pool for the stream
11042 Add method to allocate and get the multicast addresses for this stream.
11044 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11046 * docs/libs/gst-rtsp-server-sections.txt:
11047 * gst/rtsp-server/rtsp-media.c:
11048 * gst/rtsp-server/rtsp-media.h:
11049 media: remove MTU property
11050 It is a stream property
11052 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11054 * gst/rtsp-server/rtsp-client.c:
11055 client: set blocksize only on stream
11056 Set the blocksize only on the current stream.
11058 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11060 * gst/rtsp-server/rtsp-stream.c:
11061 stream: share src and sink sockets
11062 the allocated socket is in the used-socket property, not socket.
11064 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11066 * gst/rtsp-server/rtsp-address-pool.c:
11067 * gst/rtsp-server/rtsp-address-pool.h:
11068 * gst/rtsp-server/rtsp-client.c:
11069 * gst/rtsp-server/rtsp-session-media.c:
11070 * gst/rtsp-server/rtsp-session-media.h:
11071 * gst/rtsp-server/rtsp-stream-transport.c:
11072 * gst/rtsp-server/rtsp-stream-transport.h:
11073 * tests/check/gst/addresspool.c:
11074 rtsp: make address-pool return an address object
11075 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
11076 store more info in the structure and allows us to more easily return the address
11077 to the right pool when no longer needed.
11078 Pass the address to the StreamTransport so that we can return it to the pool
11079 when the stream transport is freed or changed.
11081 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11083 * examples/Makefile.am:
11084 * examples/test-multicast.c:
11085 examples: add multicast example
11086 Show how to set up the multicast address pool so that media can be
11087 server with multicast.
11089 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11091 * gst/rtsp-server/rtsp-client.c:
11092 * gst/rtsp-server/rtsp-media-factory.c:
11093 * gst/rtsp-server/rtsp-media-factory.h:
11094 * gst/rtsp-server/rtsp-media.c:
11095 * gst/rtsp-server/rtsp-media.h:
11096 rtsp: use AddressPool
11097 Remove the multicast_group property.
11098 Use the configured addresspool to allocate multicast addresses.
11100 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11102 * gst/rtsp-server/rtsp-address-pool.c:
11103 * gst/rtsp-server/rtsp-address-pool.h:
11104 address-pool: add clear method
11106 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11108 * gst/rtsp-server/rtsp-address-pool.c:
11109 address-pool: small cleanups
11111 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11113 * tests/check/Makefile.am:
11114 * tests/check/gst/addresspool.c:
11115 tests: add addresspool unit test
11117 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11119 * gst/rtsp-server/Makefile.am:
11120 * gst/rtsp-server/rtsp-address-pool.c:
11121 * gst/rtsp-server/rtsp-address-pool.h:
11122 address-pool: add object to manage multicast addresses
11123 Make an object that can manage a rage of multicast addresses and ports.
11125 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11127 * gst/rtsp-server/rtsp-server.c:
11128 server: set default max-threads property
11130 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11132 * gst/rtsp-server/rtsp-media.c:
11133 media: wait for concurrent _prepare
11134 If a prepare is busy, wait for the result.
11136 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11138 * gst/rtsp-server/rtsp-media.c:
11139 media: add lock around message handler
11140 We don't want to dispatch messages while we are still processing the result of
11143 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11145 * gst/rtsp-server/rtsp-media.c:
11146 * gst/rtsp-server/rtsp-media.h:
11147 media: add lock to protect state changes
11149 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11151 * gst/rtsp-server/rtsp-stream.c:
11152 * gst/rtsp-server/rtsp-stream.h:
11153 stream: add locking
11155 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11157 * gst/rtsp-server/rtsp-stream-transport.c:
11158 * gst/rtsp-server/rtsp-stream-transport.h:
11159 * gst/rtsp-server/rtsp-stream.c:
11160 stream-transport: add keep-alive method
11162 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11164 * gst/rtsp-server/rtsp-stream-transport.c:
11165 * gst/rtsp-server/rtsp-stream-transport.h:
11166 * gst/rtsp-server/rtsp-stream.c:
11167 stream-transport: add method to handle RTP/RTCP
11168 Call new methods instead of poking into the structures directly.
11170 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11172 * gst/rtsp-server/rtsp-session-media.c:
11173 * gst/rtsp-server/rtsp-session-media.h:
11174 session-media: add locking
11176 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11178 * gst/rtsp-server/rtsp-session.c:
11179 * gst/rtsp-server/rtsp-session.h:
11180 session: add locking
11182 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11184 * gst/rtsp-server/rtsp-server.c:
11185 server: free old socket
11187 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11189 * gst/rtsp-server/rtsp-media-mapping.c:
11190 * gst/rtsp-server/rtsp-media-mapping.h:
11191 mapping: add locking
11193 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11195 * gst/rtsp-server/rtsp-media-factory.c:
11196 media-factory: add locking
11198 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11200 * gst/rtsp-server/rtsp-auth.c:
11201 * gst/rtsp-server/rtsp-auth.h:
11204 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11206 * gst/rtsp-server/rtsp-server.c:
11207 * gst/rtsp-server/rtsp-server.h:
11208 server: add max-thread property
11210 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11212 * gst/rtsp-server/rtsp-server.c:
11213 * gst/rtsp-server/rtsp-server.h:
11214 server: use a threadpool for the mainloops
11216 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11218 * gst/rtsp-server/rtsp-client.c:
11219 * gst/rtsp-server/rtsp-client.h:
11220 client: rename method
11221 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
11222 don't really create the client from the socket, we use the socket for the
11225 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11227 * gst/rtsp-server/rtsp-client.c:
11228 * gst/rtsp-server/rtsp-client.h:
11229 * gst/rtsp-server/rtsp-server.c:
11230 server: rework maincontext handling in clients
11231 Make a separate method to attach a client to a MainContext.
11232 Let the server decide in what GMainContext the client will operate and give this
11233 context to the client in attach. Then the server can later decide to use a
11234 separate thread for each client or just use the mainthread.
11236 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11238 * gst/rtsp-server/rtsp-client.c:
11239 * gst/rtsp-server/rtsp-session.c:
11240 * gst/rtsp-server/rtsp-session.h:
11241 session: move session header code in session object
11243 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
11247 * examples/test-auth.c:
11248 * examples/test-launch.c:
11249 * examples/test-mp4.c:
11250 * examples/test-ogg.c:
11251 * examples/test-readme.c:
11252 * examples/test-sdp.c:
11253 * examples/test-uri.c:
11254 * examples/test-video.c:
11255 * gst/rtsp-server/rtsp-auth.c:
11256 * gst/rtsp-server/rtsp-auth.h:
11257 * gst/rtsp-server/rtsp-client.c:
11258 * gst/rtsp-server/rtsp-client.h:
11259 * gst/rtsp-server/rtsp-media-factory-uri.c:
11260 * gst/rtsp-server/rtsp-media-factory-uri.h:
11261 * gst/rtsp-server/rtsp-media-factory.c:
11262 * gst/rtsp-server/rtsp-media-factory.h:
11263 * gst/rtsp-server/rtsp-media-mapping.c:
11264 * gst/rtsp-server/rtsp-media-mapping.h:
11265 * gst/rtsp-server/rtsp-media.c:
11266 * gst/rtsp-server/rtsp-media.h:
11267 * gst/rtsp-server/rtsp-params.c:
11268 * gst/rtsp-server/rtsp-params.h:
11269 * gst/rtsp-server/rtsp-sdp.c:
11270 * gst/rtsp-server/rtsp-sdp.h:
11271 * gst/rtsp-server/rtsp-server.c:
11272 * gst/rtsp-server/rtsp-server.h:
11273 * gst/rtsp-server/rtsp-session-media.c:
11274 * gst/rtsp-server/rtsp-session-media.h:
11275 * gst/rtsp-server/rtsp-session-pool.c:
11276 * gst/rtsp-server/rtsp-session-pool.h:
11277 * gst/rtsp-server/rtsp-session.c:
11278 * gst/rtsp-server/rtsp-session.h:
11279 * gst/rtsp-server/rtsp-stream-transport.c:
11280 * gst/rtsp-server/rtsp-stream-transport.h:
11281 * gst/rtsp-server/rtsp-stream.c:
11282 * gst/rtsp-server/rtsp-stream.h:
11283 * tests/check/gst/rtspserver.c:
11284 * tests/test-cleanup.c:
11287 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11289 * gst/rtsp-server/rtsp-media.c:
11290 * gst/rtsp-server/rtsp-session-media.c:
11291 * gst/rtsp-server/rtsp-session.c:
11292 rtsp-server: added annotations to indicate type of ownership transfer of return values
11293 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11295 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11298 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11300 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11303 * bindings/Makefile.am:
11304 * bindings/vala/Makefile.am:
11305 * bindings/vala/gst-rtsp-server-0.10.deps:
11306 * bindings/vala/gst-rtsp-server-0.10.vapi:
11307 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11308 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11309 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11310 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11311 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11313 bindings: remove vala bindings
11314 They'll be reunited with the other GStreamer bindings
11315 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11317 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11319 * gst/rtsp-server/rtsp-client.c:
11320 * gst/rtsp-server/rtsp-session-media.c:
11321 * gst/rtsp-server/rtsp-session-media.h:
11322 * gst/rtsp-server/rtsp-stream-transport.c:
11323 * gst/rtsp-server/rtsp-stream-transport.h:
11324 rtsp: only create transport when needed
11325 Only create the StreamTransport when configured.
11327 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11329 * gst/rtsp-server/rtsp-client.c:
11330 client: small cleanup
11332 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11334 * gst/rtsp-server/rtsp-client.c:
11335 * gst/rtsp-server/rtsp-client.h:
11336 * gst/rtsp-server/rtsp-stream-transport.c:
11337 * gst/rtsp-server/rtsp-stream-transport.h:
11338 rtsp: refactor configuration of transport
11339 Move the configuration of the transport to a place where it makes
11342 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11344 * gst/rtsp-server/rtsp-client.c:
11345 client: refactor transport parsing
11347 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11349 * gst/rtsp-server/rtsp-client.c:
11350 client: refuse to change the MTU on shared media
11351 If we change the MTU of chared media, it changes for all clients.
11352 We don't want to set the MTU to something large for clients that
11355 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11357 * examples/test-mp4.c:
11358 * gst/rtsp-server/rtsp-media.c:
11359 small fixes to docs and debug
11361 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11363 * gst/rtsp-server/rtsp-stream.c:
11364 stream: transports must already have been removed
11366 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11368 * gst/rtsp-server/rtsp-media.c:
11369 * gst/rtsp-server/rtsp-stream.c:
11370 * gst/rtsp-server/rtsp-stream.h:
11371 stream: improve join and leave of the pipeline
11373 Do the cleanup properly
11376 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11378 * gst/rtsp-server/rtsp-media.c:
11379 media: move unprepare below default implementation
11380 Makes it easier to find the default implementation
11382 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11384 * gst/rtsp-server/rtsp-media.c:
11385 media: signal unprepared when we actually finish
11387 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11389 * gst/rtsp-server/rtsp-media.c:
11390 media: no need to unlock, unprepare does that when needed
11392 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11394 * docs/libs/gst-rtsp-server-sections.txt:
11395 * gst/rtsp-server/rtsp-media-factory.h:
11396 * gst/rtsp-server/rtsp-media-mapping.c:
11397 * gst/rtsp-server/rtsp-media.h:
11398 * gst/rtsp-server/rtsp-params.c:
11399 * gst/rtsp-server/rtsp-server.c:
11400 * gst/rtsp-server/rtsp-session-pool.h:
11401 * gst/rtsp-server/rtsp-session.c:
11402 * gst/rtsp-server/rtsp-session.h:
11403 * gst/rtsp-server/rtsp-stream-transport.h:
11404 * gst/rtsp-server/rtsp-stream.h:
11407 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11409 * gst/rtsp-server/rtsp-client.c:
11410 * gst/rtsp-server/rtsp-media-mapping.h:
11411 * gst/rtsp-server/rtsp-media.c:
11412 * gst/rtsp-server/rtsp-media.h:
11413 * gst/rtsp-server/rtsp-server.h:
11414 * gst/rtsp-server/rtsp-stream.c:
11415 * gst/rtsp-server/rtsp-stream.h:
11416 rtsp: fix MTU setting
11417 Fix setting of the MTU. There is no need for a vmethod.
11419 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11424 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11427 configure: bump version number after refactoring
11429 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11431 * gst/rtsp-server/Makefile.am:
11432 * gst/rtsp-server/rtsp-client.c:
11433 * gst/rtsp-server/rtsp-client.h:
11434 * gst/rtsp-server/rtsp-media-factory-uri.c:
11435 * gst/rtsp-server/rtsp-media-factory.c:
11436 * gst/rtsp-server/rtsp-media-factory.h:
11437 * gst/rtsp-server/rtsp-media.c:
11438 * gst/rtsp-server/rtsp-media.h:
11439 * gst/rtsp-server/rtsp-sdp.c:
11440 * gst/rtsp-server/rtsp-session-media.c:
11441 * gst/rtsp-server/rtsp-session-media.h:
11442 * gst/rtsp-server/rtsp-session.c:
11443 * gst/rtsp-server/rtsp-session.h:
11444 * gst/rtsp-server/rtsp-stream-transport.c:
11445 * gst/rtsp-server/rtsp-stream-transport.h:
11446 * gst/rtsp-server/rtsp-stream.c:
11447 * gst/rtsp-server/rtsp-stream.h:
11448 rtsp: massive refactoring
11449 Make GObjects from the remaining simple structures.
11450 Remove GstRTSPSessionStream, it's not needed.
11451 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11452 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11453 a GstRTSPStream should be transported to a client.
11454 Rename GstRTSPMediaFactory::get_element -> create_element because that
11455 more accurately describes what it does.
11456 Make nice methods instead of poking in the structures.
11457 Move some methods inside the relevant object source code.
11458 Use GPtrArray to store objects instead of plain arrays, it is more
11459 natural and allows us to more easily clean up.
11460 Move the allocation of udp ports to the Stream object. The Stream object
11461 contains the elements needed to stream the media to a client.
11462 Improve the prepare and unprepare methods. Unprepare should now undo
11463 everything prepare did. Improve also async unprepare when doing EOS on
11464 shutdown. Make sure we always unprepare correctly.
11466 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11468 * gst/rtsp-server/rtsp-client.c:
11469 rtsp-client: Unref server address clients connected to
11470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11472 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11474 * gst/rtsp-server/rtsp-server.c:
11475 rtsp-server: don't ref server socket if it is NULL
11476 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11477 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11479 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11481 * tests/check/Makefile.am:
11482 tests: Add libgio link dependency
11483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11485 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11487 * gst/rtsp-server/rtsp-media-mapping.c:
11488 * gst/rtsp-server/rtsp-media-mapping.h:
11489 rtsp-media-mapping: rename find_media vfunc to find_factory
11490 The virtual method and class method should have the same name
11491 so it is correctly represented in GIR file
11492 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11494 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11496 * gst/rtsp-server/rtsp-auth.c:
11497 * gst/rtsp-server/rtsp-client.c:
11498 * gst/rtsp-server/rtsp-media-factory-uri.c:
11499 * gst/rtsp-server/rtsp-media-factory.c:
11500 * gst/rtsp-server/rtsp-media-mapping.c:
11501 * gst/rtsp-server/rtsp-media.c:
11502 * gst/rtsp-server/rtsp-server.c:
11503 * gst/rtsp-server/rtsp-session-pool.c:
11504 * gst/rtsp-server/rtsp-session.c:
11505 rtsp-server: fixed comments and GIR annotations
11506 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11508 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11510 * gst/rtsp-server/rtsp-media-mapping.c:
11511 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11513 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11515 * gst/rtsp-server/rtsp-server.c:
11516 rtsp-server: allow binding on port 0 (binds on a random port)
11518 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11520 * gst/rtsp-server/rtsp-server.c:
11521 * gst/rtsp-server/rtsp-server.h:
11522 rtsp-server: add bound-port property
11523 bound-port can be used to retrieve the port number when the server is bound on
11524 port 0, which binds on a random port.
11526 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11528 * gst/rtsp-server/rtsp-media-factory.c:
11529 * gst/rtsp-server/rtsp-media-factory.h:
11530 rtsp-media-factory: make ::get_element overridable by GI bindings
11531 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11532 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11533 as the invoker for ::get_element(), making it overridable by GI generated
11536 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11538 * gst/rtsp-server/rtsp-media-factory-uri.c:
11539 rtsp-media-factory-uri: don't autoplug parsers in a loop
11540 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11543 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11545 * gst/rtsp-server/Makefile.am:
11546 Explicitly link against gio. Fix link error on mac.
11548 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11550 * gst/rtsp-server/rtsp-session.c:
11551 session: add ttl to the transport header in SETUP
11552 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11554 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11556 * gst/rtsp-server/rtsp-client.c:
11557 * gst/rtsp-server/rtsp-client.h:
11558 * gst/rtsp-server/rtsp-media.c:
11559 client: Use client transport settings for multicast if allowed.
11560 This patch makes it possible for the client to send transport settings for
11561 multicast (destination && ttl). Client settings must be explicitly allowed or
11562 the server will use its own settings.
11563 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11565 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11568 Automatic update of common submodule
11569 From 6c0b52c to 6bb6951
11571 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11573 * gst/rtsp-server/rtsp-client.c:
11574 rtsp-client: do not destroy the rtsp watch
11575 Don't destroy the client watch while dispatching. The rtsp watch is
11576 automatically destroyed after the rtsp watch function closed() has
11578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11580 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11583 Automatic update of common submodule
11584 From 4f962f7 to 6c0b52c
11586 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11588 * gst/rtsp-server/rtsp-media.c:
11589 media: fix check for seekability
11591 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11593 * gst/rtsp-server/rtsp-client.c:
11594 client: use more GIO
11595 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11597 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11599 * gst/rtsp-server/rtsp-server.c:
11600 server: remove obsolete includes
11602 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11604 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11605 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11606 be available in "on_new_ssrc". The transports are added in
11607 gst_rtsp_media_set_state when going to PLAYING state. However,
11608 "on_new_ssrc" might be called before this happens.
11609 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11611 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11613 * gst/rtsp-server/rtsp-client.c:
11614 * gst/rtsp-server/rtsp-client.h:
11615 rtsp-client: add signals for rtsp requests (fixes #683287)
11617 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11619 * gst/rtsp-server/rtsp-client.c:
11620 * gst/rtsp-server/rtsp-client.h:
11621 add new-session signal to rtsp-client (fixes #683058)
11623 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11626 Automatic update of common submodule
11627 From 668acee to 4f962f7
11629 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11631 * gst/rtsp-server/rtsp-server.c:
11632 * tests/check/gst/rtspserver.c:
11633 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11634 Do not assume that *error is set in g_socket_address_enumerator_next.
11635 Added test_bind_already_in_use unit-test.
11636 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11638 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11641 Automatic update of common submodule
11642 From 94ccf4c to 668acee
11644 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11646 * gst/rtsp-server/rtsp-client.c:
11647 * gst/rtsp-server/rtsp-client.h:
11648 rtsp-client: make create_sdp virtual method
11649 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11651 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11654 Automatic update of common submodule
11655 From 98e386f to 94ccf4c
11657 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11659 * gst/rtsp-server/rtsp-client.c:
11662 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11664 * gst/rtsp-server/rtsp-client.c:
11665 * gst/rtsp-server/rtsp-client.h:
11666 * gst/rtsp-server/rtsp-server.c:
11667 * gst/rtsp-server/rtsp-server.h:
11668 rtsp-server: use an existing socket to establish HTTP tunnel
11669 Make it possible to transfer a socket from an HTTP server to be used as
11670 an RTSP over HTTP tunnel.
11672 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11674 * gst/rtsp-server/rtsp-client.c:
11675 * gst/rtsp-server/rtsp-media.c:
11676 * gst/rtsp-server/rtsp-media.h:
11677 rtsp: Handle the blocksize parameter
11678 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11680 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11682 * tests/check/Makefile.am:
11683 * tests/check/gst/rtspserver.c:
11684 Have unit test get header from source dir, not installed dir
11685 This makes compilation of unit tests work in a build directory other
11686 than the source directory.
11687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11689 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11691 * gst/rtsp-server/rtsp-media.c:
11692 rtsp-media: update for gst_element_make_from_uri() changes
11694 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11697 * tests/Makefile.am:
11698 * tests/check/Makefile.am:
11699 * tests/check/gst/rtspserver.c:
11700 rtsp: add unit test
11701 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11703 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11705 * gst/rtsp-server/rtsp-media.c:
11706 rtsp-media: don't collect media stats when going to NULL
11707 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11709 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11711 * gst/rtsp-server/rtsp-client.c:
11712 client: don't leak transports
11714 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11716 * gst/rtsp-server/rtsp-client.c:
11717 rtsp-client: free transport on no_stream in SETUP handler
11719 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11721 * gst/rtsp-server/rtsp-client.c:
11722 rtsp-client: changed session media iteration
11723 In client_unlink_session: now don't iterate in session->medias
11724 list where items are removed by gst_rtsp_session_release_media.
11725 Instead, repeatedly remove the first item.
11727 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11729 * gst/rtsp-server/rtsp-client.c:
11730 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11731 GstRTSPSessionMedia is not a GObject type. When the
11732 GstRTSPSession is freed, it will free the media.
11734 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11736 * gst/rtsp-server/rtsp-media-factory.c:
11737 factory: plug pad leak in collect_streams
11738 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11739 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11740 will take one reference, and the other reference will otherwise
11741 give a memory leak.
11743 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11746 configure: suppress some warnings when debug is disabled
11747 Warnings about unused variables should be suppressed if core has the
11748 debug system disabled.
11749 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11751 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11753 * docs/libs/Makefile.am:
11754 docs: fix build in uninstalled setup
11755 Include gst-plugins-base libs properly.
11757 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11759 * docs/libs/gst-rtsp-server.types:
11760 docs: include headers defining rtsp-server object types
11761 Fixes compiler warnings during docs build.
11762 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11764 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11767 configure: Add warning flags for compiler when configuring
11768 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11770 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11773 Automatic update of common submodule
11774 From 03a0e57 to 98e386f
11776 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11779 Automatic update of common submodule
11780 From 1fab359 to 03a0e57
11782 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11784 * gst/rtsp-server/rtsp-client.c:
11785 client: fix GSocketAddress leak in gst_rtsp_client_accept
11786 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11788 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11791 Automatic update of common submodule
11792 From f1b5a96 to 1fab359
11794 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11797 Automatic update of common submodule
11798 From 92b7266 to f1b5a96
11800 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11803 Automatic update of common submodule
11804 From ec1c4a8 to 92b7266
11806 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11809 Automatic update of common submodule
11810 From 3429ba6 to ec1c4a8
11812 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11814 * gst/rtsp-server/rtsp-auth.c:
11815 * gst/rtsp-server/rtsp-client.c:
11816 * gst/rtsp-server/rtsp-media-factory-uri.c:
11817 * gst/rtsp-server/rtsp-server.c:
11818 rtsp: fix compiler warnings
11819 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11821 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11824 Automatic update of common submodule
11825 From dc70203 to 3429ba6
11827 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11829 * gst/rtsp-server/rtsp-client.c:
11830 * gst/rtsp-server/rtsp-media-factory.c:
11831 * gst/rtsp-server/rtsp-media-factory.h:
11832 * gst/rtsp-server/rtsp-media.c:
11833 * gst/rtsp-server/rtsp-media.h:
11834 * gst/rtsp-server/rtsp-server.c:
11835 * gst/rtsp-server/rtsp-server.h:
11836 * gst/rtsp-server/rtsp-session-pool.c:
11837 * gst/rtsp-server/rtsp-session-pool.h:
11838 rtsp-server: port to new thread API
11840 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11843 Automatic update of common submodule
11844 From 6db25be to dc70203
11846 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11848 * gst/rtsp-server/rtsp-auth.c:
11849 * gst/rtsp-server/rtsp-auth.h:
11850 * gst/rtsp-server/rtsp-client.c:
11851 rtsp-server: Fix compilation and compiler warnings
11853 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11857 * gst/rtsp-server/Makefile.am:
11858 configure: Modernize autotools setup a bit
11859 Also we now only create tar.bz2 and tar.xz tarballs.
11861 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11864 Automatic update of common submodule
11865 From 464fe15 to 6db25be
11867 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11870 Automatic update of common submodule
11871 From 7fda524 to 464fe15
11873 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11876 * docs/libs/Makefile.am:
11877 * docs/version.entities.in:
11878 * gst-rtsp.spec.in:
11879 * gst/rtsp-server/Makefile.am:
11880 * pkgconfig/Makefile.am:
11881 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11882 * pkgconfig/gstreamer-rtsp-server.pc.in:
11883 * tests/Makefile.am:
11884 rtsp-server: Update versioning
11886 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11888 Merge remote-tracking branch 'origin/0.10'
11890 gst/rtsp-server/rtsp-session-pool.c
11892 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11894 * gst/rtsp-server/rtsp-session-pool.c:
11895 rtsp-server: Don't use deprecated GLib API
11897 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11899 Replace master with 0.11
11901 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11903 Merge branch 'master' into 0.11
11905 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11907 Merge branch 'master' into 0.11
11909 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11912 A couple minor typo fixes
11914 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11916 * gst/rtsp-server/rtsp-media.c:
11917 media: fix state of the appqueue
11919 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11921 * gst/rtsp-server/rtsp-media-factory-uri.c:
11922 factory: use videoconvert
11924 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11926 * gst/rtsp-server/rtsp-media-factory-uri.c:
11927 factory: change to new style caps
11929 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11931 * gst/rtsp-server/rtsp-client.c:
11932 * gst/rtsp-server/rtsp-client.h:
11933 * gst/rtsp-server/rtsp-media-factory-uri.c:
11934 * gst/rtsp-server/rtsp-media.c:
11935 * gst/rtsp-server/rtsp-server.c:
11936 * gst/rtsp-server/rtsp-server.h:
11937 * gst/rtsp-server/rtsp-session-pool.c:
11938 rtsp-server: port to GIO
11941 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11944 configure: fix build
11946 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11949 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11950 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11952 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11955 * examples/Makefile.am:
11956 First rule of gst-rtsp-server club: don't talk about gst-phonon
11958 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11961 * pkgconfig/Makefile.am:
11962 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11963 * pkgconfig/gstreamer-rtsp-server.pc.in:
11964 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11965 For consistency with all other modules.
11967 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11969 * gst/rtsp-server/rtsp-client.c:
11970 rtsp-client: update for new map API
11972 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11975 * bindings/Makefile.am:
11976 * bindings/python/Makefile.am:
11977 * bindings/python/arg-types.py:
11978 * bindings/python/codegen/Makefile.am:
11979 * bindings/python/codegen/__init__.py:
11980 * bindings/python/codegen/argtypes.py:
11981 * bindings/python/codegen/code-coverage.py:
11982 * bindings/python/codegen/codegen.py:
11983 * bindings/python/codegen/definitions.py:
11984 * bindings/python/codegen/defsparser.py:
11985 * bindings/python/codegen/docextract.py:
11986 * bindings/python/codegen/docgen.py:
11987 * bindings/python/codegen/fileprefix.override:
11988 * bindings/python/codegen/fileprefixmodule.c:
11989 * bindings/python/codegen/h2def.py:
11990 * bindings/python/codegen/mergedefs.py:
11991 * bindings/python/codegen/mkskel.py:
11992 * bindings/python/codegen/override.py:
11993 * bindings/python/codegen/reversewrapper.py:
11994 * bindings/python/codegen/scmexpr.py:
11995 * bindings/python/rtspserver-types.defs:
11996 * bindings/python/rtspserver.defs:
11997 * bindings/python/rtspserver.override:
11998 * bindings/python/rtspservermodule.c:
11999 * bindings/python/test.py:
12001 python: remove pygst-based python bindings
12002 pygi is the future, apparently.
12004 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
12007 Automatic update of common submodule
12008 From c463bc0 to 7fda524
12010 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12013 Automatic update of common submodule
12014 From 2a59016 to c463bc0
12016 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12019 Automatic update of common submodule
12020 From 0807187 to 2a59016
12022 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12025 Automatic update of common submodule
12026 From 11f0cd5 to 0807187
12028 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12030 * examples/test-auth.c:
12031 example: update for new caps
12033 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12035 * examples/test-video.c:
12036 * gst/rtsp-server/rtsp-client.c:
12037 * gst/rtsp-server/rtsp-media-factory-uri.c:
12038 * gst/rtsp-server/rtsp-media.c:
12039 * gst/rtsp-server/rtsp-media.h:
12040 * gst/rtsp-server/rtsp-session.c:
12041 * gst/rtsp-server/rtsp-session.h:
12042 rtsp-server: port some more to 0.11
12044 Remove bufferlist stuff
12045 Update for new API.
12046 Add queue before appsink now that preroll-queue-len is gone.
12047 Update for request pad changes.
12049 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12051 Merge branch 'master' into 0.11
12053 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12055 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12056 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12057 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12059 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12061 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12062 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12063 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12065 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12067 Merge branch 'master' into 0.11
12069 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12071 * gst/rtsp-server/rtsp-media.c:
12072 * gst/rtsp-server/rtsp-media.h:
12073 media: add a seekable boolean
12074 Maintain the seekable state with a new variable instead of reusing the
12077 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
12079 * gst/rtsp-server/rtsp-media.c:
12080 Disallow seek in live media
12082 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12084 Merge branch 'master' into 0.11
12086 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
12088 * gst/rtsp-server/rtsp-server.c:
12089 #ifdef statements for windows socket creation were missing
12091 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
12094 Automatic update of common submodule
12095 From a39eb83 to 11f0cd5
12097 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
12100 Automatic update of common submodule
12101 From 605cd9a to a39eb83
12103 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12105 Merge branch 'master' into 0.11
12107 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12109 * gst/rtsp-server/rtsp-client.c:
12110 client: use method to access property
12112 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12114 * gst/rtsp-server/rtsp-media-factory.c:
12115 * gst/rtsp-server/rtsp-media-factory.h:
12116 media-factory: add protocols property
12117 Add a property to configure the allowed protocols in the media created from the
12120 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12122 * gst/rtsp-server/rtsp-media-factory.c:
12123 * gst/rtsp-server/rtsp-media-factory.h:
12124 media-factory: add media-configure signal
12125 Add signal to allow the application to configure the media after it was created
12128 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12130 * gst/rtsp-server/rtsp-client.c:
12131 client: use method to access property
12133 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12135 * gst/rtsp-server/rtsp-media-factory.c:
12136 * gst/rtsp-server/rtsp-media-factory.h:
12137 media-factory: add protocols property
12138 Add a property to configure the allowed protocols in the media created from the
12141 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12143 * gst/rtsp-server/rtsp-media-factory.c:
12144 * gst/rtsp-server/rtsp-media-factory.h:
12145 media-factory: add media-configure signal
12146 Add signal to allow the application to configure the media after it was created
12149 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12151 Merge branch 'master' into 0.11
12153 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12155 * gst/rtsp-server/rtsp-client.c:
12156 client: use media multicast group
12158 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12160 * gst/rtsp-server/rtsp-media-factory.h:
12161 * gst/rtsp-server/rtsp-server.h:
12162 * gst/rtsp-server/rtsp-session-pool.h:
12163 * gst/rtsp-server/rtsp-session.h:
12166 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12168 * gst/rtsp-server/rtsp-client.c:
12169 * gst/rtsp-server/rtsp-sdp.h:
12170 sdp: copy and free the server ip address
12171 Copy and free the server ip address to make memory management easier later.
12173 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12175 * gst/rtsp-server/rtsp-media-factory.c:
12176 media-factory: configure multicast in media
12178 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12180 * gst/rtsp-server/rtsp-media.c:
12181 * gst/rtsp-server/rtsp-media.h:
12182 media: add property for multicast group
12183 Add a property to configure the multicast group in the media.
12184 Based on patches from Marc Leeman and Robert Krakora.
12186 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12188 * gst/rtsp-server/rtsp-media-factory.c:
12189 * gst/rtsp-server/rtsp-media-factory.h:
12190 media-factory: add property for multicast group
12191 Add a property to configure the multicast group in the media factory.
12192 Based on patches from Marc Leeman and Robert Krakora.
12194 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12196 * gst/rtsp-server/rtsp-client.c:
12197 client: do configuration of transport in one place
12198 Move the configuration of the transport destination address to where we also
12199 configure the other bits.
12201 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12203 * gst/rtsp-server/rtsp-client.c:
12204 client: use media multicast group
12206 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12208 * gst/rtsp-server/rtsp-media-factory.h:
12209 * gst/rtsp-server/rtsp-server.h:
12210 * gst/rtsp-server/rtsp-session-pool.h:
12211 * gst/rtsp-server/rtsp-session.h:
12214 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12216 * gst/rtsp-server/rtsp-client.c:
12217 * gst/rtsp-server/rtsp-sdp.h:
12218 sdp: copy and free the server ip address
12219 Copy and free the server ip address to make memory management easier later.
12221 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12223 * gst/rtsp-server/rtsp-media-factory.c:
12224 media-factory: configure multicast in media
12226 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12228 * gst/rtsp-server/rtsp-media.c:
12229 * gst/rtsp-server/rtsp-media.h:
12230 media: add property for multicast group
12231 Add a property to configure the multicast group in the media.
12232 Based on patches from Marc Leeman and Robert Krakora.
12234 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12236 * gst/rtsp-server/rtsp-media-factory.c:
12237 * gst/rtsp-server/rtsp-media-factory.h:
12238 media-factory: add property for multicast group
12239 Add a property to configure the multicast group in the media factory.
12240 Based on patches from Marc Leeman and Robert Krakora.
12242 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12244 * gst/rtsp-server/rtsp-client.c:
12245 client: do configuration of transport in one place
12246 Move the configuration of the transport destination address to where we also
12247 configure the other bits.
12249 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12251 Merge branch 'master' into 0.11
12253 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12255 * gst/rtsp-server/rtsp-client.c:
12256 client: destroy pipeline on client disconnect with no prior TEARDOWN.
12257 The problem occurs when the client abruptly closes the connection without
12258 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
12259 server is where the pipeline gets torn down. Since this handler is not called,
12260 the pipeline remains and is up and running. Subsequent clients get their own
12261 pipelines and if the do not issue TEARDOWNs then those pipelines will also
12262 remain up and running. This is a resource leak.
12264 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12266 Merge branch 'master' into 0.11
12268 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
12270 * gst/rtsp-server/rtsp-media-factory.c:
12271 * gst/rtsp-server/rtsp-media-factory.h:
12272 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
12273 For example, it can be used to retrieve source elements like appsrc, in a more
12274 convenient way than subclassing get_element.
12276 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12278 Merge branch 'master' into 0.11
12280 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
12282 * gst/rtsp-server/rtsp-server.c:
12283 rtsp-server: hold on to reference while using object
12285 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12287 * gst/rtsp-server/rtsp-media.c:
12290 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12293 configure: use unstable api
12295 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12297 * gst/rtsp-server/rtsp-client.c:
12298 client: fix reference counting
12300 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12302 * gst/rtsp-server/rtsp-client.c:
12303 * gst/rtsp-server/rtsp-media.c:
12304 fix compiler warnings about unused variables
12306 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12308 * examples/test-launch.c:
12309 * examples/test-readme.c:
12310 * examples/test-uri.c:
12311 * examples/test-video.c:
12312 examples: tell rtsp uri when ready
12314 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12317 Automatic update of common submodule
12318 From 69b981f to 605cd9a
12320 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12322 * gst/rtsp-server/rtsp-client.c:
12323 client: update for buffer API change
12325 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12327 * gst/rtsp-server/Makefile.am:
12328 Makefile.am: 0.10 => @GST_MAJORMINOR@
12330 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12332 * gst/rtsp-server/rtsp-media-factory-uri.c:
12333 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12335 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12337 * gst/rtsp-server/.gitignore:
12338 .gitignore: 0.10 => 0.11
12340 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12342 * gst/rtsp-server/Makefile.am:
12343 Makefile.am: 0.10 => @GST_MAJORMINOR@
12345 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12347 Merge branch 'master' into 0.11
12349 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12352 Automatic update of common submodule
12353 From 9e5bbd5 to 69b981f
12355 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12358 Automatic update of common submodule
12359 From fd35073 to 9e5bbd5
12361 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12364 Automatic update of common submodule
12365 From 46dfcea to fd35073
12367 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12369 * gst/rtsp-server/rtsp-media-factory-uri.c:
12370 * gst/rtsp-server/rtsp-media.c:
12371 media: port to new caps API
12373 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12375 Merge branch 'master' into 0.11
12377 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12379 * bindings/vala/gst-rtsp-server-0.10.vapi:
12380 Updated Vala bindings.
12381 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12383 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12385 * gst/rtsp-server/rtsp-server.c:
12386 * gst/rtsp-server/rtsp-server.h:
12387 Add a signal for newly connected clients.
12388 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12390 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12392 * bindings/python/rtspserver.override:
12393 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12395 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12397 * gst/rtsp-server/Makefile.am:
12398 * gst/rtsp-server/rtsp-client.c:
12399 * gst/rtsp-server/rtsp-funnel.c:
12400 * gst/rtsp-server/rtsp-funnel.h:
12401 * gst/rtsp-server/rtsp-media.c:
12402 rtsp-server: port to 0.11
12404 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12409 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12411 Merge branch 'master' into 0.11
12416 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12419 Automatic update of common submodule
12420 From c3cafe1 to 46dfcea
12422 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12424 * bindings/python/Makefile.am:
12425 * bindings/python/rtspserver.defs:
12426 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12428 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12430 * bindings/python/arg-types.py:
12431 python bindings: add GstRTSPUrlParam
12432 Needed to implement MediaFactory virtual proxies
12434 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12436 * bindings/python/arg-types.py:
12437 python bindings: fix returning GstRTSPUrl types
12439 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12441 * bindings/python/arg-types.py:
12442 python bindings: add arg type for GstRTSPUrl
12444 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12446 * bindings/python/rtspserver.defs:
12447 python bindings: fix the definition of MediaFactory.collect_stream
12449 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12452 Automatic update of common submodule
12453 From 1ccbe09 to c3cafe1
12455 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12458 Automatic update of common submodule
12459 From 193b717 to 1ccbe09
12461 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12464 Automatic update of common submodule
12465 From b77e2bf to 193b717
12467 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12470 build: Include lcov.mak to allow test coverage report generation
12472 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12475 Automatic update of common submodule
12476 From d8814b6 to b77e2bf
12478 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12481 Automatic update of common submodule
12482 From 6aaa286 to d8814b6
12484 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12487 Automatic update of common submodule
12488 From 6aec6b9 to 6aaa286
12490 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12493 autogen: wingo signed comment
12495 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12497 * gst/rtsp-server/rtsp-session-pool.c:
12498 session: use full charset for RTSP session ID
12499 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12500 session ID more difficult.
12501 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12503 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12505 * gst/rtsp-server/Makefile.am:
12506 rtsp-server: Don't install the funnel header
12508 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12511 Automatic update of common submodule
12512 From 1de7f6a to 6aec6b9
12514 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12517 configure: require core/base 0.10.31
12518 Needed at least for gst_plugin_feature_rank_compare_func().
12520 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12523 Automatic update of common submodule
12524 From f94d739 to 1de7f6a
12526 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12528 * gst/rtsp-server/rtsp-media.c:
12529 media: remove more unused code
12531 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12533 * gst/rtsp-server/rtsp-media.c:
12534 * gst/rtsp-server/rtsp-media.h:
12535 media: remove duplicate filtering
12536 Remove the duplicate filtering code now that we have a released -good version.
12537 Give a warning instead.
12539 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12541 * gst/rtsp-server/rtsp-media-factory.c:
12542 * gst/rtsp-server/rtsp-media.c:
12543 media: fix default buffer size
12545 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12547 * gst/rtsp-server/rtsp-media-factory.c:
12548 * gst/rtsp-server/rtsp-media-factory.h:
12549 media-factory: add property to configure the buffer-size
12550 Add a property to configure the kernel UDP buffer size.
12552 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12554 * gst/rtsp-server/rtsp-media.c:
12555 * gst/rtsp-server/rtsp-media.h:
12556 media: add property to configure kernel buffer sizes
12557 Add a property to configure the kernel UDP buffer size.
12559 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12562 configure: set PYGOBJECT_REQ before using it
12563 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12565 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12567 * docs/Makefile.am:
12568 docs: recursive into sub-directories on 'make upload'
12570 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12572 * docs/libs/gst-rtsp-server-docs.sgml:
12573 * docs/version.entities.in:
12574 docs: mention full version these docs are for, not just major-minor
12576 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12579 back to development
12581 === release 0.10.8 ===
12583 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12588 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12590 * gst/rtsp-server/rtsp-server.c:
12591 rtsp-server: clarify docs a little
12593 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12595 * gst/rtsp-server/rtsp-media.c:
12596 media: init debug category before starting thread
12598 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12600 * gst/rtsp-server/rtsp-auth.c:
12601 auth: add realm to make it more spec compliant
12603 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12605 * gst/rtsp-server/rtsp-server.c:
12606 * gst/rtsp-server/rtsp-server.h:
12607 server: add locking
12609 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12611 * examples/test-video.c:
12612 example: improve example docs a little
12614 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12616 * gst/rtsp-server/rtsp-server.c:
12617 server: ensure the watch has a ref to the server
12619 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12621 * gst/rtsp-server/rtsp-server.c:
12622 server: simpify channel function
12624 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12626 * gst/rtsp-server/rtsp-server.c:
12627 * gst/rtsp-server/rtsp-server.h:
12628 server: simplify management of channel and source
12629 We don't need to keep around the channel and source objects. Let the mainloop
12630 and the source manage the source and channel respectively.
12632 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12638 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12640 * tests/.gitignore:
12641 * tests/Makefile.am:
12642 * tests/test-cleanup.c:
12643 tests: add tests directory and cleanup test
12645 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12647 * gst/rtsp-server/rtsp-media-factory-uri.c:
12648 * gst/rtsp-server/rtsp-media-factory.c:
12649 * gst/rtsp-server/rtsp-media-mapping.c:
12650 * gst/rtsp-server/rtsp-media.c:
12651 * gst/rtsp-server/rtsp-session-pool.c:
12652 * gst/rtsp-server/rtsp-session.c:
12653 server: improve debugging in various objects
12655 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12657 * gst/rtsp-server/rtsp-server.c:
12658 server: chain up to the parent finalize
12660 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12662 * bindings/python/rtspserver-types.defs:
12663 * bindings/python/rtspserver.defs:
12664 * bindings/python/rtspserver.override:
12665 * bindings/python/test.py:
12666 gst-rtsp-server: update python bindings
12668 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12670 * gst/rtsp-server/rtsp-client.c:
12671 client: use the response from the clientstate
12672 Create the response object only once and store in the client state.
12673 Make all methods use the state response,
12675 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12677 * gst/rtsp-server/rtsp-server.c:
12678 server: use signal to keep track of clients
12679 Keep track of all the clients that the server creates and remove them when they
12680 fire the 'closed' signal.
12682 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12684 * gst/rtsp-server/rtsp-client.c:
12685 * gst/rtsp-server/rtsp-client.h:
12686 client: emit signal when closing
12688 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12690 * examples/.gitignore:
12691 * examples/Makefile.am:
12692 * examples/test-auth.c:
12693 * examples/test-video.c:
12694 * gst/rtsp-server/rtsp-auth.c:
12695 * gst/rtsp-server/rtsp-auth.h:
12696 * gst/rtsp-server/rtsp-client.c:
12697 * gst/rtsp-server/rtsp-media-factory.c:
12698 * gst/rtsp-server/rtsp-media.c:
12699 * gst/rtsp-server/rtsp-media.h:
12700 * gst/rtsp-server/rtsp-session-pool.h:
12701 * gst/rtsp-server/rtsp-session.h:
12702 media: enable per factory authorisations
12703 Allow for adding a GstRTSPAuth on the factory and media level and check
12704 permissions when accessing the factory.
12705 Add hints to the auth methods for future more fine grained authorisation.
12706 Add example application for per factory authentication.
12708 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12710 * gst/rtsp-server/rtsp-auth.c:
12711 * gst/rtsp-server/rtsp-auth.h:
12712 * gst/rtsp-server/rtsp-client.c:
12713 * gst/rtsp-server/rtsp-client.h:
12714 * gst/rtsp-server/rtsp-params.c:
12715 * gst/rtsp-server/rtsp-params.h:
12716 rtsp-server: Pass ClientState structure arround
12717 Pass the collected information for the ongoing request in a GstRTSPClientState
12718 structure that we can then pass around to simplify the method arguments. This
12719 will also be handy when we implement logging functionality.
12721 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12723 * gst/rtsp-server/rtsp-media-factory.c:
12724 * gst/rtsp-server/rtsp-media-factory.h:
12725 media-factory: add methods to configure authorisation
12727 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12729 * gst/rtsp-server/rtsp-client.c:
12730 client: unref auth in finalize
12732 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12734 * gst/rtsp-server/rtsp-server.c:
12735 server: unref auth in finalize
12737 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12739 * docs/libs/gst-rtsp-server-docs.sgml:
12740 * docs/libs/gst-rtsp-server-sections.txt:
12741 * docs/libs/gst-rtsp-server.types:
12742 docs: add more docs
12744 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12746 * gst/rtsp-server/rtsp-server.c:
12747 * gst/rtsp-server/rtsp-server.h:
12748 server: separate create and accept
12749 Create separate create and accept methods so that subclasses can create custom
12751 Configure the server in the client object and prepare for keeping track of
12754 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12756 * gst/rtsp-server/rtsp-client.c:
12757 * gst/rtsp-server/rtsp-client.h:
12758 client: add support for setting the server.
12759 Add support for keeping a ref to the server that started this client
12762 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12764 * gst/rtsp-server/rtsp-auth.c:
12765 auth: fix memleak and add some docs
12766 Fix a memleak of the basic auth token.
12767 Add docs for the helper function
12769 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12771 * gst/rtsp-server/rtsp-auth.c:
12772 * gst/rtsp-server/rtsp-auth.h:
12773 * gst/rtsp-server/rtsp-client.c:
12774 client: delegate setup of auth to the manager
12775 Delegate the configuration of the authentication tokens to the manager object
12778 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12780 * examples/test-video.c:
12781 * gst/rtsp-server/Makefile.am:
12782 * gst/rtsp-server/rtsp-auth.c:
12783 * gst/rtsp-server/rtsp-auth.h:
12784 * gst/rtsp-server/rtsp-client.c:
12785 * gst/rtsp-server/rtsp-client.h:
12786 * gst/rtsp-server/rtsp-server.c:
12787 * gst/rtsp-server/rtsp-server.h:
12788 auth: add authentication object
12789 Add an object that can check the authorization of requests.
12790 Implement basic authentication.
12791 Add example authentication to test-video
12793 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12795 * gst/rtsp-server/rtsp-server.c:
12796 * gst/rtsp-server/rtsp-server.h:
12797 server: move includes back
12798 the includes are needed for sockaddr_in.
12800 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12802 * gst/rtsp-server/rtsp-client.c:
12803 * gst/rtsp-server/rtsp-client.h:
12804 * gst/rtsp-server/rtsp-server.c:
12805 * gst/rtsp-server/rtsp-server.h:
12806 rtsp: move network includes where they are needed
12808 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12810 * gst/rtsp-server/rtsp-media.h:
12811 rtsp-media.h: Minor corrections in comments.
12814 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12817 Automatic update of common submodule
12818 From e572c87 to f94d739
12820 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12824 * docs/libs/.gitignore:
12825 * examples/.gitignore:
12826 * gst/rtsp-server/.gitignore:
12829 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12831 * docs/libs/Makefile.am:
12832 docs: We don't build ps/pdf for API reference docs
12834 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12837 Automatic update of common submodule
12838 From ccbaa85 to e572c87
12840 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12843 Automatic update of common submodule
12844 From 46445ad to ccbaa85
12846 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12848 * gst/rtsp-server/Makefile.am:
12849 * gst/rtsp-server/rtsp-funnel.c:
12850 * gst/rtsp-server/rtsp-funnel.h:
12851 * gst/rtsp-server/rtsp-media.c:
12852 funnel: rename fsfunnel to rtspfunnel
12853 Rename the funnel to avoid conflicts with the farsight one.
12855 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12857 * gst/rtsp-server/Makefile.am:
12858 * gst/rtsp-server/fs-funnel.c:
12859 * gst/rtsp-server/fs-funnel.h:
12860 * gst/rtsp-server/rtsp-media.c:
12861 rtsp-media: add and use fsfunnel
12862 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12863 select-all property that we need.
12865 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12867 * gst/rtsp-server/Makefile.am:
12868 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12869 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12870 for the g-ir-compiler, rather than just assuming the env var has
12873 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12880 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12882 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12885 * gst/rtsp-server/Makefile.am:
12886 gobject-introspection: fix g-i build for uninstalled setup
12887 Requires gst-plugins-base git (> 0.10.31.2).
12889 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12891 * examples/test-uri.c:
12892 examples: add some more options and comments
12894 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12896 * gst/rtsp-server/rtsp-media-factory-uri.c:
12897 factory-uri: use right property type
12899 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12901 * gst/rtsp-server/rtsp-media-factory-uri.c:
12902 factory-uri: attempt to configure buffer-lists
12903 Attempt to configure buffer lists in the payloader for improved performance.
12905 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12907 * gst/rtsp-server/rtsp-media.c:
12908 media: attempt to configure bigger UDP buffers
12909 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12910 send buffers with high bitrate streams.
12912 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12914 * gst/rtsp-server/rtsp-client.c:
12915 client: use the socket length from getsockname
12916 Use the length returned by getsockname to perform the getnameinfo call because
12917 the size can depend on the socket type and platform.
12920 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12922 * docs/libs/gst-rtsp-server-docs.sgml:
12923 * docs/libs/gst-rtsp-server-sections.txt:
12924 docs: add uri factory to the docs
12926 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12928 * gst/rtsp-server/rtsp-client.c:
12929 * gst/rtsp-server/rtsp-media.h:
12932 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12934 * gst/rtsp-server/rtsp-client.c:
12935 * gst/rtsp-server/rtsp-media.c:
12936 * gst/rtsp-server/rtsp-media.h:
12937 * gst/rtsp-server/rtsp-session.c:
12938 * gst/rtsp-server/rtsp-session.h:
12939 rtsp-server: add support for buffer lists
12940 Add support for sending bufferlists received from appsink.
12943 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12945 * gst/rtsp-server/rtsp-client.c:
12946 * gst/rtsp-server/rtsp-media.c:
12947 * gst/rtsp-server/rtsp-media.h:
12948 * gst/rtsp-server/rtsp-sdp.c:
12949 media: make method to retrieve the play range
12950 Make a method to retrieve the playback range so that we can conditionally create
12951 a different range for the SDP and the PLAY requests.
12953 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12955 * gst/rtsp-server/rtsp-media.c:
12956 * gst/rtsp-server/rtsp-media.h:
12957 media: add signal to notify of state changes
12959 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12961 * gst/rtsp-server/rtsp-client.h:
12962 client: cleanup headers
12964 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12966 * gst/rtsp-server/rtsp-client.c:
12969 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12971 * gst/rtsp-server/rtsp-media-factory-uri.c:
12972 * gst/rtsp-server/rtsp-media-factory-uri.h:
12973 factory-uri: add support for gstpay
12974 Add an option to prefer gstpay over decoder + raw payloader.
12976 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12978 * gst/rtsp-server/rtsp-media-factory-uri.c:
12979 * gst/rtsp-server/rtsp-media-factory-uri.h:
12980 factory-uri: rework the autoplugger.
12981 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12984 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12986 * gst/rtsp-server/rtsp-media-factory-uri.c:
12987 factory-uri: use better factory filter
12988 Make better payloader filter based on autoplug rank and RTP use case.
12990 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12993 Automatic update of common submodule
12994 From 169462a to 46445ad
12996 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12998 * gst/rtsp-server/rtsp-server.c:
12999 server: set SO_REUSEADDR before bind
13000 Set the SO_REUSEADDR _before_ bind() to make it actually work.
13002 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13004 * gst/rtsp-server/rtsp-media.c:
13005 * gst/rtsp-server/rtsp-media.h:
13006 media: emit prepared signal when prepared
13007 Make a 'prepared' signal and emit it when we successfully prepared the element.
13008 This signal can be used to configure the media object after it has been prepared
13011 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
13014 Automatic update of common submodule
13015 From 011bcc8 to 169462a
13017 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
13019 python an optional dependency
13020 * configure.ac: Move up valgrind and g-i checks. Make the python
13021 dependency optional, as it was before.
13023 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13025 Merge branch 'master' into 0.11
13030 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13032 * gst/rtsp-server/rtsp-media.c:
13033 media: update range when active clients changed
13034 When we changed the number of active clients, update the current range
13035 information because we want the second client connecting to a shared resource
13036 continue from where the stream currently.
13038 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13040 * gst/rtsp-server/rtsp-media-factory-uri.c:
13041 * gst/rtsp-server/rtsp-media-factory-uri.h:
13042 factory-uri: add colorspace and fix pt
13043 Rework the way we pass data to the autoplugger.
13044 When we have raw caps, plug a converter element to make pluggin to raw
13045 payloaders more successful.
13046 Make sure all dynamically plugged payloaders have a unique payload types.
13048 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13050 * examples/Makefile.am:
13051 * examples/test-uri.c:
13052 example: add example of the uri factory
13054 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13056 * gst/rtsp-server/Makefile.am:
13057 * gst/rtsp-server/rtsp-media-factory-uri.c:
13058 * gst/rtsp-server/rtsp-media-factory-uri.h:
13059 * gst/rtsp-server/rtsp-server.h:
13060 factory-uri: add a factory to stream any URI
13061 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
13064 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13066 * gst/rtsp-server/rtsp-media.c:
13067 * gst/rtsp-server/rtsp-media.h:
13068 media: ignore spurious ASYNC_DONE messages
13069 When we are dynamically adding pads, the addition of the udpsrc elements will
13070 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
13071 the real ASYNC_DONE when everything is prerolled.
13073 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13075 * gst/rtsp-server/rtsp-media-factory.c:
13076 * gst/rtsp-server/rtsp-media-factory.h:
13077 media-factory: make lock macro
13079 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
13081 * gst/rtsp-server/rtsp-client.c:
13082 rtsp-server: Remove unused variable and dead assignment
13084 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
13086 * examples/test-launch.c:
13087 * examples/test-mp4.c:
13088 * examples/test-ogg.c:
13089 * examples/test-readme.c:
13090 * examples/test-sdp.c:
13091 * examples/test-video.c:
13092 examples: Run gst-indent
13094 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
13096 * gst/rtsp-server/rtsp-client.c:
13097 * gst/rtsp-server/rtsp-media-factory.c:
13098 * gst/rtsp-server/rtsp-media-mapping.c:
13099 * gst/rtsp-server/rtsp-media.c:
13100 * gst/rtsp-server/rtsp-params.c:
13101 * gst/rtsp-server/rtsp-sdp.c:
13102 * gst/rtsp-server/rtsp-server.c:
13103 * gst/rtsp-server/rtsp-session-pool.c:
13104 * gst/rtsp-server/rtsp-session.c:
13105 rtsp-server: Run gst-indent
13106 Since it wasn't using the upstream common previously, there was no
13107 indentation check before commiting.
13109 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
13111 * gst/rtsp-server/rtsp-media-mapping.h:
13112 * gst/rtsp-server/rtsp-media.c:
13113 * gst/rtsp-server/rtsp-media.h:
13114 * gst/rtsp-server/rtsp-sdp.c:
13115 * gst/rtsp-server/rtsp-session-pool.h:
13116 * gst/rtsp-server/rtsp-session.c:
13117 * gst/rtsp-server/rtsp-session.h:
13118 rtsp-server: Some more doc fixups
13120 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13123 Makefile: Add cruft-cleaning support
13125 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13129 * docs/Makefile.am:
13130 * docs/libs/Makefile.am:
13131 * docs/libs/gst-rtsp-server-docs.sgml:
13132 * docs/libs/gst-rtsp-server-sections.txt:
13133 * docs/libs/gst-rtsp-server.types:
13134 * docs/version.entities.in:
13135 docs: Add gtk-doc build system
13137 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13139 * gst/rtsp-server/Makefile.am:
13140 Makefile.am: Use standard GIR make behaviour
13142 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13146 autogen/configure: Bring more in sync to standard gst module behaviour
13148 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13150 * gst/rtsp-server/rtsp-media.c:
13151 media: warn and fail when gstrtpbin is not found
13153 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13156 configure: open 0.11 branch
13158 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
13162 Add common submodule
13164 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
13166 * common/ChangeLog:
13167 * common/Makefile.am:
13168 * common/c-to-xml.py:
13169 * common/check.mak:
13170 * common/coverage/coverage-report-entry.pl:
13171 * common/coverage/coverage-report.pl:
13172 * common/coverage/coverage-report.xsl:
13173 * common/coverage/lcov.mak:
13174 * common/gettext.patch:
13175 * common/glib-gen.mak:
13176 * common/gst-autogen.sh:
13177 * common/gst-xmlinspect.py:
13179 * common/gstdoc-scangobj:
13180 * common/gtk-doc-plugins.mak:
13181 * common/gtk-doc.mak:
13182 * common/m4/.gitignore:
13183 * common/m4/Makefile.am:
13184 * common/m4/README:
13185 * common/m4/as-ac-expand.m4:
13186 * common/m4/as-auto-alt.m4:
13187 * common/m4/as-compiler-flag.m4:
13188 * common/m4/as-compiler.m4:
13189 * common/m4/as-docbook.m4:
13190 * common/m4/as-libtool-tags.m4:
13191 * common/m4/as-libtool.m4:
13192 * common/m4/as-python.m4:
13193 * common/m4/as-scrub-include.m4:
13194 * common/m4/as-version.m4:
13195 * common/m4/ax_create_stdint_h.m4:
13196 * common/m4/check.m4:
13197 * common/m4/glib-gettext.m4:
13198 * common/m4/gst-arch.m4:
13199 * common/m4/gst-args.m4:
13200 * common/m4/gst-check.m4:
13201 * common/m4/gst-debuginfo.m4:
13202 * common/m4/gst-default.m4:
13203 * common/m4/gst-doc.m4:
13204 * common/m4/gst-error.m4:
13205 * common/m4/gst-feature.m4:
13206 * common/m4/gst-function.m4:
13207 * common/m4/gst-gettext.m4:
13208 * common/m4/gst-glib2.m4:
13209 * common/m4/gst-libxml2.m4:
13210 * common/m4/gst-plugindir.m4:
13211 * common/m4/gst-valgrind.m4:
13212 * common/m4/gtk-doc.m4:
13213 * common/m4/introspection.m4:
13214 * common/m4/pkg.m4:
13215 * common/mangle-tmpl.py:
13216 * common/plugins.xsl:
13218 * common/release.mak:
13219 * common/scangobj-merge.py:
13220 * common/upload.mak:
13221 common: Remove static version
13223 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
13225 * common/m4/introspection.m4:
13226 Update introspection.m4 to match usage
13228 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13232 Remove old stuff from the README
13234 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13237 back to development
13239 === release 0.10.7 ===
13241 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13246 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13248 * examples/test-ogg.c:
13249 test-ogg: remove parsers
13250 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
13251 buffers with timestamps. Using the parsers also seems to break things.
13253 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13255 * bindings/vala/gst-rtsp-server-0.10.vapi:
13256 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13257 Updated Vala bindings
13259 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13261 * common/m4/introspection.m4:
13263 * gst/rtsp-server/Makefile.am:
13264 Added initial gobject-introspection support
13266 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13268 * gst/rtsp-server/rtsp-media-factory.c:
13269 media-factory: don't use host for shared hash key
13270 When we generate the key to share made between connections, don't include the
13271 host used to connect so that we can share media even if between clients that
13272 connected with localhost and ones with the ip address.
13274 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13276 * bindings/vala/Makefile.am:
13277 build: fix distcheck
13279 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13281 * bindings/vala/gst-rtsp-server-0.10.vapi:
13282 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13283 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13284 Update Vala bindings
13286 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13288 * bindings/vala/Makefile.am:
13290 Fix configure checks and installation location for Vala bindings
13293 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13296 back to development
13298 === release 0.10.6 ===
13300 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13303 configure: release 0.10.6
13305 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13307 * gst/rtsp-server/rtsp-media.c:
13308 media: help the compiler a little
13310 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13312 * gst/rtsp-server/rtsp-media.c:
13313 * gst/rtsp-server/rtsp-media.h:
13314 * gst/rtsp-server/rtsp-session.c:
13315 media: cleanup media transport before freeing
13316 Cleanup the media transport data before freeing. In particular, remove the qdata
13317 from the rtpsource object.
13319 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13321 * gst/rtsp-server/rtsp-media-factory.c:
13322 * gst/rtsp-server/rtsp-media-factory.h:
13323 * gst/rtsp-server/rtsp-media.c:
13324 * gst/rtsp-server/rtsp-media.h:
13325 media-factory: add eos-shutdown property
13326 Add an eos-shutdown property that will send an EOS to the pipeline before
13327 shutting it down. This allows for nice cleanup in case of a muxer.
13330 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13332 * gst/rtsp-server/rtsp-media.c:
13333 * gst/rtsp-server/rtsp-media.h:
13334 media: use multiudpsink send-duplicates when we can
13335 If we have a new enough multiudpsink with the send-duplicates property, use this
13336 instead of doing our own filtering. Our custom filtering code should eventually
13337 be removed when we can depend on a released -good.
13339 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13341 * gst/rtsp-server/rtsp-media.c:
13342 media: don't leak destinations
13343 Refactor and cleanup the destinations array when the stream is destroyed.
13345 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13347 * gst/rtsp-server/rtsp-media.c:
13348 * gst/rtsp-server/rtsp-media.h:
13349 media: don't add udp addresses multiple times
13350 Keep track of the udp addresses we added to udpsink and never add the same udp
13351 destination twice. This avoids duplicate packets when using multicast.
13353 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13355 * gst/rtsp-server/rtsp-server.c:
13356 server: disable use of SO_LINGER
13357 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13358 server close()s the connection.
13360 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13362 * gst/rtsp-server/rtsp-server.c:
13363 server: use 5 second linger period in SO_LINGER
13364 Wait 5 seconds before clearing the send buffers and reseting the connection with
13365 the client when we do a close. This should be enough time to get the message to
13369 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13371 * gst/rtsp-server/rtsp-server.c:
13372 server: use SO_LINGER
13373 SO_LINGER on the socket will make sure that any pending data on the socket is
13374 flushed ASAP and that the socket connection is reset. This makes sure that the
13375 socket can be reused immediately.
13378 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13381 README: add blurb about shared media factories
13383 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13385 * gst/rtsp-server/rtsp-media.c:
13386 Add stdlib.h for atoi()
13388 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13390 * bindings/python/Makefile.am:
13391 * bindings/vala/Makefile.am:
13392 build: distcheck fixes
13393 Fix 'make distcheck', somewhat (it still fails because it tries to
13394 install files into /usr/share/vala/vapi/ irrespective of the
13395 configured prefix).
13397 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13400 configure: bump core/base requirements to released version
13401 Makes things less confusing for people.
13403 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13406 configure: fail if GStreamer core/base requirements are not met
13408 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13410 * gst/rtsp-server/rtsp-client.c:
13411 client: improve client cleanups
13412 Make sure the session does not timeout when using TCP. We need to do this
13413 because quicktime player does not send RTCP for some reason in tunneled
13415 Refactor some cleanup code.
13418 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13420 * gst/rtsp-server/rtsp-session.c:
13421 * gst/rtsp-server/rtsp-session.h:
13422 session: add support for prevent session timeouts
13423 Add an atomix counter to prevent session timeouts when we are, for example,
13424 streaming over TCP.
13426 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13428 * gst/rtsp-server/rtsp-client.c:
13429 client: fix unlink on session timeouts
13430 When our session times out, make sure we unlink all streams in this
13432 Remove the tunnelid when closing the connection.
13434 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13436 * gst/rtsp-server/rtsp-session.c:
13437 session: small cleanups
13439 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13441 * gst/rtsp-server/rtsp-client.c:
13442 client: handle lost_tunnel callbacks
13443 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13444 hashtable so that we can reuse it for when the client reopens the POST
13446 Close the connection after a TEARDOWN.
13447 Make sure or watchid is cleared when the watch is removed.
13450 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13452 * gst/rtsp-server/rtsp-client.c:
13453 * gst/rtsp-server/rtsp-media.c:
13454 * gst/rtsp-server/rtsp-sdp.c:
13455 rtsp-server: add more support for multicast
13457 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13460 * gst/rtsp-server/rtsp-media.c:
13461 * gst/rtsp-server/rtsp-media.h:
13462 media: allow configuration of allowed lower transport
13464 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13466 * gst/rtsp-server/rtsp-client.h:
13467 * gst/rtsp-server/rtsp-media.c:
13468 * gst/rtsp-server/rtsp-media.h:
13469 * gst/rtsp-server/rtsp-sdp.c:
13470 * gst/rtsp-server/rtsp-sdp.h:
13471 * gst/rtsp-server/rtsp-server.c:
13472 rtsp: keep track of server ip and ipv6
13473 Keep track of how the client connected to the server and setup the udp ports
13474 with the same protocol.
13475 Copy the server ip address in the SDP so that clients can send RTCP back to
13478 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13480 * gst/rtsp-server/rtsp-session.c:
13483 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13485 * gst/rtsp-server/rtsp-client.c:
13486 client: use right size for malloc
13488 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13490 * gst/rtsp-server/rtsp-server.c:
13491 server: comment ipv6 server listening address
13493 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13495 * gst/rtsp-server/rtsp-media.c:
13496 media: allow for ipv6 sockets
13498 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13500 * gst/rtsp-server/rtsp-server.c:
13501 * gst/rtsp-server/rtsp-server.h:
13502 server: rework server part
13503 Allow setting a bind address, make sure we can deal with ipv6.
13504 Remove the port property and change with the service property.
13506 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13508 * gst/rtsp-server/rtsp-media.h:
13509 media: update comments a little
13511 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13513 * gst/rtsp-server/rtsp-client.c:
13514 client: make content-base better
13515 Use the URI formatting functions to make a content-base. Also make sure that
13516 there is a trailing / at the end.
13518 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13520 * gst/rtsp-server/rtsp-client.c:
13521 client: guard against invalid paths
13523 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13525 * examples/test-video.c:
13526 test: catch server bind errors
13528 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13530 * gst/rtsp-server/rtsp-media.c:
13531 rtspmedia: emit "unprepared" if _prepare fails.
13532 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13533 media object is removed from its factory's cache.
13535 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13537 * gst/rtsp-server/rtsp-media.c:
13538 media: collect media position when seek completes
13540 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13542 * gst/rtsp-server/rtsp-client.c:
13543 client: call unlink_streams in client finalize
13546 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13548 * gst/rtsp-server/rtsp-media.c:
13549 media: limit the time to wait to something huge
13550 Avoid waiting forever but limit the timeout to 20 seconds.
13552 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13554 * gst/rtsp-server/rtsp-sdp.c:
13555 sdp: reindent and check for prepared status
13557 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13559 * gst/rtsp-server/rtsp-media.c:
13560 * gst/rtsp-server/rtsp-media.h:
13561 * gst/rtsp-server/rtsp-session.c:
13562 media: avoid doing _get_state() for state changes
13563 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13564 until the media is prerolled or in error. This avoids doing a blocking call of
13565 gst_element_get_state() that can cause lockups when there is an error.
13568 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13570 * gst/rtsp-server/rtsp-media.c:
13573 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13575 * gst/rtsp-server/rtsp-media-factory.c:
13576 media-factory: better error handling
13577 Improve the error handling a bit.
13579 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13581 * gst/rtsp-server/rtsp-client.c:
13582 client: rework transport parsing
13583 Rework the transport parsing code so that we can ignore transports we don't
13584 support instead of just picking the first one we can parse.
13585 Configure a (for now hardcoded) destination for multicast transports.
13587 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13589 * gst/rtsp-server/rtsp-media.c:
13590 media: set multicast sink parameters
13591 Disable loop and automatic multicast join on the udpsink elements.
13592 Add some more debug info.
13593 Reset some state variables in the right place.
13594 Use the right port numbers for multicast.
13596 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13598 * gst/rtsp-server/rtsp-session.c:
13599 session: handle transport setup correctly
13600 Handle UDP, MCAST and TCP transport negotiation more correctly.
13601 Store the server session SSRC in the transport.
13603 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13605 * gst/rtsp-server/rtsp-client.c:
13606 rtsp-client: implement error_full
13607 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13610 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13613 * gst/rtsp-server/rtsp-client.c:
13614 * gst/rtsp-server/rtsp-server.c:
13615 docs: update docs and comments
13617 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13619 * gst/rtsp-server/rtsp-sdp.c:
13620 sdp: make server work better when behind a proxy
13622 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13624 * gst/rtsp-server/rtsp-client.c:
13625 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13627 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13629 * gst/rtsp-server/rtsp-client.c:
13630 * gst/rtsp-server/rtsp-media-factory.c:
13631 * gst/rtsp-server/rtsp-media-mapping.c:
13632 * gst/rtsp-server/rtsp-media.c:
13633 * gst/rtsp-server/rtsp-server.c:
13634 * gst/rtsp-server/rtsp-session-pool.c:
13635 * gst/rtsp-server/rtsp-session.c:
13636 Use GStreamer's debugging subsystem
13638 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13640 * gst/rtsp-server/rtsp-media-factory.c:
13641 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13643 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13646 back to development
13648 === release 0.10.5 ===
13650 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13655 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13658 configure: bump required versions
13660 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13662 * gst/rtsp-server/rtsp-client.c:
13663 client: call weak-unref on client->sessions from finalize
13666 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13668 * gst/rtsp-server/rtsp-media.c:
13669 media: Fixed crasher where caps got unref'ed too often
13671 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13674 * pkgconfig/.gitignore:
13675 * pkgconfig/Makefile.am:
13676 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13677 Added pkg-config file to use gst-rtsp-server uninstalled
13679 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13681 * gst/rtsp-server/rtsp-media.c:
13682 media: add some docs
13684 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13686 * gst/rtsp-server/rtsp-client.c:
13687 rtsp: Use gst_rtsp_watch_send_message().
13688 Use gst_rtsp_watch_send_message() since the old API which used
13689 gst_rtsp_watch_queue_message() has been deprecated.
13691 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13694 back to development
13696 === release 0.10.4 ===
13698 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13703 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13705 * gst/rtsp-server/rtsp-client.c:
13706 * gst/rtsp-server/rtsp-session.c:
13707 * gst/rtsp-server/rtsp-session.h:
13708 rtsp: allocate channels in TCP mode
13709 When the client does not provide us with channels in TCP mode, allocate channels
13712 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13714 * gst/rtsp-server/rtsp-client.c:
13715 client: don't crash when tunnelid is missing
13716 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13717 don't crash but return an error response to the client.
13720 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13722 * bindings/vala/gst-rtsp-server-0.10.vapi:
13723 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13724 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13725 bindings: update vala bindings with new method
13727 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13729 * gst/rtsp-server/rtsp-session-pool.c:
13730 * gst/rtsp-server/rtsp-session-pool.h:
13731 sessionpool: add function to filter sessions
13732 Add generic function to retrieve/remove sessions.
13734 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13737 configure: bump core/base requirements to release
13739 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13741 * gst/rtsp-server/rtsp-media.c:
13742 media: fix indentation
13744 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13746 * gst/rtsp-server/rtsp-media.c:
13747 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13749 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13751 * gst/rtsp-server/rtsp-media.c:
13752 set state and remove elements of media in for loop
13754 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13756 * bindings/vala/gst-rtsp-server-0.10.vapi:
13757 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13758 Added gst_rtsp_media_remove_elements function to Vala bindings
13760 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13762 * gst/rtsp-server/rtsp-media.c:
13763 * gst/rtsp-server/rtsp-media.h:
13764 Added gst_rtsp_media_remove_elements function
13766 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13768 * gst/rtsp-server/rtsp-media.c:
13769 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13771 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13773 * bindings/vala/gst-rtsp-server-0.10.vapi:
13774 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13775 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13776 Updated Vala bindings
13778 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13780 * gst/rtsp-server/rtsp-media.c:
13781 * gst/rtsp-server/rtsp-media.h:
13782 Added vmethod unprepare to GstRTSPMedia
13783 The default implementation sets the state of the pipeline to GST_STATE_NULL
13785 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13787 * gst/rtsp-server/rtsp-media-factory.c:
13788 * gst/rtsp-server/rtsp-media-factory.h:
13789 Made collect_streams function public
13791 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13793 * gst/rtsp-server/rtsp-media-factory.c:
13794 * gst/rtsp-server/rtsp-media-factory.h:
13795 * gst/rtsp-server/rtsp-media.c:
13796 Added vmethod create_pipeline to GstRTSPMediaFactory
13797 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13799 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13801 * gst/rtsp-server/rtsp-client.c:
13802 client: use g_source_destroy()
13803 We need to use g_source_destroy() because we might have added the source to a
13804 different main context than the default one.
13806 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13808 * gst/rtsp-server/Makefile.am:
13809 * gst/rtsp-server/rtsp-client.c:
13810 * gst/rtsp-server/rtsp-params.c:
13811 * gst/rtsp-server/rtsp-params.h:
13812 rtsp: prepare for handling GET/SET_PARAMETER
13813 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13815 Fix return codes of handlers.
13817 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13819 * gst/rtsp-server/rtsp-media.c:
13820 media: don't leak session pads
13822 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13824 * gst/rtsp-server/rtsp-media.c:
13825 media: clean up the messages a bit
13827 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13829 * gst/rtsp-server/rtsp-sdp.c:
13830 sdp: warn and skip streams without media
13832 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13834 * bindings/vala/gst-rtsp-server-0.10.vapi:
13835 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13836 vala: Fixed typo in header file of RTSPMediaStream
13838 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13840 * gst/rtsp-server/rtsp-media.c:
13842 Fix a debug message
13843 Make dumping RTCP stats configurable
13845 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13847 * gst/rtsp-server/rtsp-media.c:
13848 media: be less verbose and leak less
13850 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13852 * gst/rtsp-server/rtsp-media.c:
13853 media: don't leak the destination address
13855 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13857 * gst/rtsp-server/rtsp-client.c:
13858 * gst/rtsp-server/rtsp-media.c:
13859 * gst/rtsp-server/rtsp-media.h:
13860 * gst/rtsp-server/rtsp-session.c:
13861 * gst/rtsp-server/rtsp-session.h:
13862 rtsp: use RTCP to keep the session alive
13863 Use the RTCP rtcp-from stats field to find the associated session and use this
13864 to keep the session alive.
13866 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13868 * gst/rtsp-server/rtsp-session.c:
13869 session: add 5sec to the real session timeout
13870 Allow the session to live 5sec longer before really timing out. This should give
13871 clients some extra time to keep the session active.
13873 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13875 * gst/rtsp-server/rtsp-client.c:
13876 client: replay OK to GET/SET_PARAMETER
13877 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13878 so that we return OK for those requests.
13880 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13882 * gst/rtsp-server/rtsp-media.c:
13883 * gst/rtsp-server/rtsp-media.h:
13884 media: keep track of active transports
13885 Keep track of which transport is active to avoid closing the connection too
13887 Remove the destination transport also when going to NULL.
13888 Print some stats about the SDES and other RTCP messages we receive from the
13891 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13893 * examples/.gitignore:
13894 * examples/Makefile.am:
13895 * examples/test-sdp.c:
13896 example: add SDP relay example
13898 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13900 * gst/rtsp-server/rtsp-media.c:
13901 media: also count active TCP connections
13903 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13905 * gst/rtsp-server/rtsp-media-factory.c:
13906 * gst/rtsp-server/rtsp-media.c:
13907 * gst/rtsp-server/rtsp-media.h:
13908 rtsp: add support for dynamic elements
13909 Add support for dynamic elements.
13910 Don't set live pipelines back to paused.
13912 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13914 * gst/rtsp-server/rtsp-sdp.c:
13915 sdp: don't add encoding name when absent in caps
13917 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13919 * gst/rtsp-server/rtsp-client.c:
13920 client: warn when we can't do RTP-Info
13922 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13924 * gst/rtsp-server/rtsp-media-factory.c:
13925 factory: factor out the stream construction
13927 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13929 * gst/rtsp-server/rtsp-client.c:
13930 client: only add RTP-Info when we have the info
13931 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13934 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13937 back to development
13939 === release 0.10.3 ===
13941 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13945 - Fixes a bug where it put the wrong verion in pkgconfig
13946 - Link RTP and RTCP sources
13948 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13950 * gst/rtsp-server/rtsp-media.c:
13951 * gst/rtsp-server/rtsp-media.h:
13952 media: link the RTP udpsrc to the session manager
13953 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13954 shut down when the client sends a packet to open firewalls.
13956 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13958 * pkgconfig/gst-rtsp-server.pc.in:
13959 Don't use hard-coded version number in pkg-config file
13961 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13964 back to development
13966 === release 0.10.2 ===
13968 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13973 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13976 * common/m4/.gitignore:
13977 * examples/.gitignore:
13978 * pkgconfig/.gitignore:
13979 add some .gitignore files
13981 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13983 * gst/rtsp-server/rtsp-media.c:
13984 media: seek to key frames
13986 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13988 * gst/rtsp-server/rtsp-media.c:
13989 media: emit the unprepared signal by id
13990 Emit the unprepared signal by id instead of name and set the media as
13993 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13995 * gst/rtsp-server/rtsp-media.c:
13996 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13998 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14000 * gst/rtsp-server/rtsp-server.c:
14001 Added finalize function to GstRTPSPServer to unref session pool and media mapping
14003 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14005 * bindings/vala/gst-rtsp-server-0.10.vapi:
14006 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14007 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14008 Updated vala bindings
14010 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14012 * gst/rtsp-server/Makefile.am:
14013 * gst/rtsp-server/rtsp-client.c:
14014 * gst/rtsp-server/rtsp-media.c:
14015 server: use appsink and appsrc with the API
14016 Use the appsink/appsrc API instead of the signals for higher
14019 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14021 * examples/test-ogg.c:
14022 tests: set the payload type correctly
14024 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14026 * gst/rtsp-server/rtsp-media-factory.c:
14027 factory: connect to the unprepare signal
14028 Connect to the unprepare signal for non-reusable media so that we can remove
14029 them from the cache.
14031 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14033 * gst/rtsp-server/rtsp-media.c:
14034 * gst/rtsp-server/rtsp-media.h:
14035 media: add signal to notify of unprepare
14037 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14039 * gst/rtsp-server/rtsp-media.c:
14040 * gst/rtsp-server/rtsp-media.h:
14041 media: more work on making the media shared
14042 Add a reusable flag to medias, indicating that they can be reused after a state
14046 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14048 * examples/test-readme.c:
14049 examples: mark the example as shared for testing
14051 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14053 * gst/rtsp-server/rtsp-media.c:
14054 * gst/rtsp-server/rtsp-media.h:
14055 client: support shared media
14056 Always perform the state actions even if the target state of the pipeline is
14057 already correct, we still want to add/remove the transports when we are dealing
14059 Keep a counter of the number of active transports for a media so that we can use
14060 this to perform a state change when needed.
14061 Perform a state change of the pipeline only when the first transport was added
14062 or when there are no active transports.
14064 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14066 * gst/rtsp-server/rtsp-client.c:
14067 client: fix refcounting crasher
14068 Don't need to remove the weak refs in the finalize methods, they are already
14069 removed in the dispose.
14070 Don't register the callback with a DestroyNofity.
14072 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14074 * gst/rtsp-server/rtsp-client.c:
14075 Fix rtsp client refcount management in TCP mode.
14076 Don't unref a client ref we never had. Fixes an unref
14077 of an already-free client object after a client
14078 teardown request for me.
14080 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14082 * gst/rtsp-server/rtsp-session.c:
14083 docs: fix typo in API docs
14085 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14087 * gst/rtsp-server/rtsp-media.c:
14088 More seeking fixes.
14089 Keep the udp sources in playing even if we go to paused. unlock the sources when
14091 Add some more debug info.
14092 Only seek when we need to.
14093 Keep track of the position when we go to paused.
14095 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14097 * gst/rtsp-server/rtsp-client.c:
14098 * gst/rtsp-server/rtsp-media.c:
14099 * gst/rtsp-server/rtsp-media.h:
14100 Add beginnings of seeking.
14101 Parse the Range header and perform a seek on the pipeline for the requested
14102 position. It's disabled currently until I figure out what's going wrong.
14104 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14106 * gst/rtsp-server/rtsp-client.c:
14107 allow pause requests for now.
14110 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14112 * gst/rtsp-server/rtsp-client.c:
14113 Remove weak ref on the session in teardown
14114 We need to remove our weakref from the session when we do a teardown because
14115 else we close the TCP connection prematurely.
14117 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14119 * gst/rtsp-server/rtsp-client.c:
14120 * gst/rtsp-server/rtsp-client.h:
14121 * gst/rtsp-server/rtsp-session-pool.c:
14122 Do some more session cleanup
14123 Make session timeout kill the TCP connection that currently watches the
14125 Remove the client timeout property.
14127 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14129 * gst/rtsp-server/rtsp-client.c:
14130 * gst/rtsp-server/rtsp-client.h:
14131 * gst/rtsp-server/rtsp-media.c:
14132 * gst/rtsp-server/rtsp-media.h:
14133 * gst/rtsp-server/rtsp-server.c:
14134 * gst/rtsp-server/rtsp-session.c:
14135 * gst/rtsp-server/rtsp-session.h:
14137 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
14140 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14142 * examples/Makefile.am:
14143 * examples/test-launch.c:
14144 Add example server that takes launch lines
14145 Add an example server that streams any -launch line.
14147 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14149 * examples/test-readme.c:
14150 * gst/rtsp-server/rtsp-client.c:
14151 * gst/rtsp-server/rtsp-media.c:
14152 * gst/rtsp-server/rtsp-media.h:
14153 Add support for live streams
14154 Add support for live streams and ranges
14155 Start on handling TCP data transfer.
14157 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14159 * gst/rtsp-server/rtsp-media.c:
14160 Free the pipeline before other things
14163 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14165 * gst/rtsp-server/rtsp-client.c:
14166 Only free the pending tunnel if there is one
14169 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14171 * gst/rtsp-server/rtsp-client.c:
14172 * gst/rtsp-server/rtsp-client.h:
14173 * gst/rtsp-server/rtsp-media.c:
14174 rtsp-server: Add support for tunneling
14175 Add support for tunneling over HTTP.
14176 Use new connection methods to retrieve the url.
14177 Dispatch messages based on the message type instead of blindly
14178 assuming it's always a request.
14179 Keep track of the watch id so that we can remove it later.
14180 Set the media pipeline to NULL before unreffing the pipeline.
14182 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14184 * gst/rtsp-server/rtsp-client.c:
14185 * gst/rtsp-server/rtsp-client.h:
14186 Fix for channel -> watch rename in gstreamer
14187 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
14189 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14191 * gst/rtsp-server/rtsp-client.c:
14192 * gst/rtsp-server/rtsp-client.h:
14194 Use the async RTSP channels instead of spawning a new thread for each client.
14195 If a sessionid is specified in a request, fail if we don't have the session.
14197 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14199 * gst/rtsp-server/rtsp-media.c:
14200 Add better debug info
14201 Add some better debug info.
14203 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14205 * examples/test-video.c:
14207 Add support for session timeouts in the example.
14209 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14211 * gst/rtsp-server/rtsp-session-pool.c:
14212 * gst/rtsp-server/rtsp-session-pool.h:
14213 Pass GTimeVal around for performance reasons
14214 Get the current time only once and pass it around so that sessions don't have to
14215 get the current time anymore.
14216 Add experimental support for a GSource that dispatches when the session needs to
14219 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14221 * gst/rtsp-server/rtsp-session.c:
14222 * gst/rtsp-server/rtsp-session.h:
14223 Add better support for session timeouts
14224 Add a method to request the number of milliseconds when a session will timeout.
14226 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14228 * gst/rtsp-server/rtsp-media.c:
14229 * gst/rtsp-server/rtsp-media.h:
14230 Add suport for RTP manager monitoring
14231 Add the first stage in monitoring the rtp manager.
14232 Make sure we don't update the state to something we don't want.
14234 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14236 * gst/rtsp-server/rtsp-client.c:
14237 Add support for session keepalive
14238 Get and update the session timeout for all requests. get the session as early as
14241 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14243 * gst/rtsp-server/rtsp-media-factory.h:
14244 * gst/rtsp-server/rtsp-media.c:
14245 * gst/rtsp-server/rtsp-media.h:
14246 Handle media bus messages
14247 Handle media bus messages in a custom mainloop and dispatch them to the
14248 RTSPMedia objects. Let the default implementation handle some common messages.
14250 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14252 * gst/rtsp-server/rtsp-client.c:
14253 * gst/rtsp-server/rtsp-session-pool.c:
14254 * gst/rtsp-server/rtsp-session.c:
14255 Some more session timeout handling
14256 Move the session header setting code to a central place so that we always add
14257 the timeout parameter too.
14258 Handle timeouts by running the session cleanup code.
14259 Stop media before cleaning up.
14261 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14263 * gst/rtsp-server/rtsp-client.c:
14264 * gst/rtsp-server/rtsp-client.h:
14265 Add timeout property
14266 Add a timeout property ot the client and make the other properties into GObject
14269 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14271 * gst/rtsp-server/rtsp-session-pool.c:
14272 Use getters and setters in property code
14273 Use the getters and setters for the timeout property instead of locking
14276 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14278 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
14280 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14282 * gst/rtsp-server/rtsp-session-pool.c:
14283 * gst/rtsp-server/rtsp-session-pool.h:
14284 * gst/rtsp-server/rtsp-session.c:
14285 * gst/rtsp-server/rtsp-session.h:
14286 Add more timeout stuff
14287 Add method to check if a session is expired.
14288 Add method to perform cleanup on a session pool.
14290 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14292 * gst/rtsp-server/rtsp-client.c:
14293 * gst/rtsp-server/rtsp-session-pool.c:
14294 * gst/rtsp-server/rtsp-session-pool.h:
14295 * gst/rtsp-server/rtsp-session.c:
14296 * gst/rtsp-server/rtsp-session.h:
14297 Add beginnings of session timeouts and limits
14298 Add the timeout value to the Session header for unusual timeout values.
14299 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14300 limit on the amount of retry we do after a sessionid collision.
14301 Add properties to the sessionid and the timeout of a session. Keep track of
14302 creation time and last access time for sessions.
14304 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14306 * gst/rtsp-server/rtsp-client.c:
14307 * gst/rtsp-server/rtsp-media.c:
14308 * gst/rtsp-server/rtsp-media.h:
14309 * gst/rtsp-server/rtsp-sdp.c:
14310 * gst/rtsp-server/rtsp-session-pool.c:
14311 * gst/rtsp-server/rtsp-session.c:
14312 * gst/rtsp-server/rtsp-session.h:
14313 Cleanup of sessions and more
14314 Fix the refcounting of media and sessions in the client. Properly clean up the
14315 session data when the client performs a teardown.
14316 Add Server header to responses.
14317 Allow for multiple uri setups in one session.
14318 Add Range header to the PLAY response and add the range attribute to the SDP
14320 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14321 give the ownership of the sessionid to the session object.
14323 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14325 * gst/rtsp-server/rtsp-server.c:
14326 * gst/rtsp-server/rtsp-server.h:
14328 Rename the 'server_port' variable to simply 'port'.
14330 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14333 * gst/rtsp-server/rtsp-client.c:
14334 * gst/rtsp-server/rtsp-media.c:
14335 * gst/rtsp-server/rtsp-media.h:
14336 * gst/rtsp-server/rtsp-session.c:
14337 * gst/rtsp-server/rtsp-session.h:
14338 Rework the way we handle transports for streams
14339 Make the media accept an array of transports for the streams that we have
14340 configured for the play/pause requests.
14341 Implement server states for a client and its media.
14342 Require 0.10.22.1 (git HEAD) of gstreamer.
14344 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14346 * gst/rtsp-server/rtsp-client.c:
14347 * gst/rtsp-server/rtsp-media-factory.c:
14348 Drop const from functions dealing with urls
14349 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14350 have the right const in them.
14352 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14354 * gst/rtsp-server/rtsp-client.c:
14355 * gst/rtsp-server/rtsp-media.c:
14356 * gst/rtsp-server/rtsp-sdp.c:
14360 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14362 * gst/rtsp-server/rtsp-client.c:
14363 * gst/rtsp-server/rtsp-media-factory.c:
14364 * gst/rtsp-server/rtsp-media.c:
14365 * gst/rtsp-server/rtsp-media.h:
14367 Don't keep a reference to the GstRTSPMedia in the stream.
14368 Free more things when freeing the GstRTSPMedia.
14370 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14373 * gst/rtsp-server/rtsp-media-factory.c:
14374 * gst/rtsp-server/rtsp-media-factory.h:
14375 * gst/rtsp-server/rtsp-media.c:
14376 * gst/rtsp-server/rtsp-media.h:
14377 * gst/rtsp-server/rtsp-server.c:
14378 * gst/rtsp-server/rtsp-server.h:
14379 More docs and small cleanups
14380 Add some more docs and update the README
14381 Cleanup some method names.
14382 Remove an unneeded idx field in the GstRTSPMediaStream
14384 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14387 * examples/Makefile.am:
14388 * examples/test-readme.c:
14389 Add a README and more example code
14390 Add a README file that contains a small introduction on how to use the server
14391 along with the example code explained in the readme.
14393 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14395 * gst/rtsp-server/rtsp-media.c:
14396 * gst/rtsp-server/rtsp-server.c:
14397 Fix some leaks and change default port
14398 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14399 we finished the initial preroll. If we keep them locked, setting the pipeline to
14400 NULL will not stop and clean up the sources correctly.
14401 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14403 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14405 * gst/rtsp-server/rtsp-session.c:
14406 * gst/rtsp-server/rtsp-session.h:
14407 Cleanups to the session object
14408 Remove some unneeded variables in the session state of a stream such as the
14409 owner media and the server transport.
14410 Get the configuration of a media stream in a session based on the media_stream
14411 in the original object instead of our cached index.
14412 Free more data in the finalize method.
14414 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14416 * gst/rtsp-server/rtsp-client.c:
14417 * gst/rtsp-server/rtsp-client.h:
14418 Cleanups and reuse media from DESCRIBE
14419 Handle thread create errors.
14420 Rename some internal methods to better match what they actually do.
14421 Handle misconfiguration of session_pool and media_mapping gracefully.
14422 Cache the DESCRIBE media and uri in the client connection and reuse them when
14423 we receive a SETUP request in the same connection for the same uri.
14424 Cleanup the client connection object.
14426 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14428 * gst/rtsp-server/rtsp-media-factory.c:
14429 * gst/rtsp-server/rtsp-media-factory.h:
14430 * gst/rtsp-server/rtsp-media.c:
14431 * gst/rtsp-server/rtsp-media.h:
14432 Add shared properties to media and factory
14433 Add the shared property to media.
14434 Implement some simple caching in the factory depending on if the media is shared
14437 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14439 * gst/rtsp-server/rtsp-client.c:
14440 Add a little comment
14441 Add some comment about the content-base header.
14443 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14445 * examples/Makefile.am:
14446 * examples/test-mp4.c:
14447 * examples/test-ogg.c:
14448 * examples/test-video.c:
14449 * gst/rtsp-server/Makefile.am:
14450 * gst/rtsp-server/rtsp-client.c:
14451 * gst/rtsp-server/rtsp-client.h:
14452 * gst/rtsp-server/rtsp-media-factory.c:
14453 * gst/rtsp-server/rtsp-media-factory.h:
14454 * gst/rtsp-server/rtsp-media.c:
14455 * gst/rtsp-server/rtsp-media.h:
14456 * gst/rtsp-server/rtsp-sdp.c:
14457 * gst/rtsp-server/rtsp-sdp.h:
14458 * gst/rtsp-server/rtsp-server.c:
14459 * gst/rtsp-server/rtsp-server.h:
14460 * gst/rtsp-server/rtsp-session.c:
14461 * gst/rtsp-server/rtsp-session.h:
14462 Reorganize things, prepare for media sharing
14463 Added various other test server examples
14464 Move the SDP message generation to a separate helper.
14465 Refactor common code for finding the session.
14466 Add content-base for realplayer compatibility
14467 Clean up request uris before processing for better vlc compatibility.
14468 Move prerolling and pipeline construction to the RTSPMedia object.
14469 Use multiudpsink for future pipeline reuse.
14471 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14474 Back to development
14477 === release 0.10.1 ===
14479 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14482 Make 0.10.1 release
14485 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14487 * bindings/vala/Makefile.am:
14489 Add more directories and files to the dist.
14491 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14493 * bindings/python/Makefile.am:
14494 * bindings/python/rtspserver.override:
14495 Fixed compile error of python bindings
14497 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14499 * bindings/vala/gst-rtsp-server-0.10.vapi:
14500 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14501 Marked values as nullable accordingly
14503 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14505 * bindings/vala/gst-rtsp-server-0.10.vapi:
14506 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14507 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14508 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14509 Updated Vala bindings
14511 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14513 * gst/rtsp-server/rtsp-client.c:
14514 * gst/rtsp-server/rtsp-media-mapping.c:
14515 * gst/rtsp-server/rtsp-media-mapping.h:
14516 * gst/rtsp-server/rtsp-media.h:
14517 * gst/rtsp-server/rtsp-session-pool.h:
14518 Cleanups and doc updates
14519 Add some more documentation and do some minor cleanups here and there.
14521 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14523 * gst/rtsp-server/rtsp-client.c:
14524 * gst/rtsp-server/rtsp-media-factory.c:
14525 * gst/rtsp-server/rtsp-media-factory.h:
14526 * gst/rtsp-server/rtsp-media.c:
14527 * gst/rtsp-server/rtsp-media.h:
14528 * gst/rtsp-server/rtsp-session.c:
14529 * gst/rtsp-server/rtsp-session.h:
14531 Rename GstRTSPMediaBin to GstRTSPMedia
14532 Parse the request url into a GstRTSPUri object and pass this object to the
14533 various handlers and methods that require the uri.
14535 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14539 Add some more docs and remove some old code from the example.
14541 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14543 * gst/rtsp-server/rtsp-client.c:
14544 Handle state change failures better
14545 Handle state change failures better when changing the state of the pipeline to
14548 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14550 * gst/rtsp-server/rtsp-media-factory.c:
14551 * gst/rtsp-server/rtsp-media-factory.h:
14552 Make element creation more extendible
14553 Add get_element vmethod to the default MediaFactory so that subclasses can just
14554 override that method and still use the default logic for making a MediaBin from
14557 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14560 * gst/rtsp-server/Makefile.am:
14561 * gst/rtsp-server/rtsp-client.c:
14562 * gst/rtsp-server/rtsp-client.h:
14563 * gst/rtsp-server/rtsp-media-factory.c:
14564 * gst/rtsp-server/rtsp-media-factory.h:
14565 * gst/rtsp-server/rtsp-media-mapping.c:
14566 * gst/rtsp-server/rtsp-media-mapping.h:
14567 * gst/rtsp-server/rtsp-media.c:
14568 * gst/rtsp-server/rtsp-media.h:
14569 * gst/rtsp-server/rtsp-server.c:
14570 * gst/rtsp-server/rtsp-server.h:
14571 * gst/rtsp-server/rtsp-session.c:
14572 * gst/rtsp-server/rtsp-session.h:
14573 Make the server handle arbitrary pipelines
14574 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14575 The GstMediaBin object has a handle to a bin with elements and to a list of
14576 GstMediaStream objects that this bin produces.
14577 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14578 with methods to register and remove those mappings.
14579 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14580 used by the server instance.
14581 Modify the example application so that it shows how to create custom pipelines
14582 attached to a specific mount point.
14583 Various misc cleanps.
14585 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14587 * gst/rtsp-server/rtsp-server.c:
14588 * gst/rtsp-server/rtsp-server.h:
14589 Allow setting a custom media factory for a server
14591 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14593 * gst/rtsp-server/rtsp-client.c:
14594 * gst/rtsp-server/rtsp-client.h:
14595 Allow setting a custom media factory for a client.
14597 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14599 * gst/rtsp-server/Makefile.am:
14600 Add Makefile entry for the media factory
14602 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14604 * gst/rtsp-server/rtsp-media-factory.c:
14605 * gst/rtsp-server/rtsp-media-factory.h:
14606 Add media factory to map urls to media pipeline objects.
14608 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14610 * gst/rtsp-server/rtsp-media.c:
14611 * gst/rtsp-server/rtsp-media.h:
14612 Add comments. Remove unused field
14614 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14616 * gst/rtsp-server/rtsp-session-pool.c:
14617 * gst/rtsp-server/rtsp-session-pool.h:
14618 Allow custom session pools to override the session id allocation algorithms Add some comments.
14620 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14622 * gst/rtsp-server/rtsp-session.h:
14625 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14627 * gst/rtsp-server/rtsp-client.c:
14628 * gst/rtsp-server/rtsp-client.h:
14629 Move the connection code in one place Add some comments
14631 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14633 * gst/rtsp-server/rtsp-server.c:
14634 * gst/rtsp-server/rtsp-server.h:
14635 Make vmethod to create and accept new clients. Add some docs.
14637 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14639 * gst/rtsp-server/rtsp-server.c:
14640 * gst/rtsp-server/rtsp-server.h:
14641 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14643 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14645 * gst/rtsp-server/rtsp-client.c:
14646 * gst/rtsp-server/rtsp-client.h:
14647 Name the parameters more appropriately.
14649 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14651 * gst/rtsp-server/rtsp-session-pool.c:
14652 Do some more cleanup of the session pool.
14654 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14656 * gst/rtsp-server/Makefile.am:
14657 * gst/rtsp-server/rtsp-client.c:
14658 Check if return value of gst_rtsp_session_get_media is not NULL
14660 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14662 * gst/rtsp-server/Makefile.am:
14663 Install rtsp-session and rtsp-session-pool headers
14665 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14670 * bindings/python/Makefile.am:
14671 * bindings/python/arg-types.py:
14672 * bindings/python/codegen/Makefile.am:
14673 * bindings/python/codegen/__init__.py:
14674 * bindings/python/codegen/argtypes.py:
14675 * bindings/python/codegen/code-coverage.py:
14676 * bindings/python/codegen/codegen.py:
14677 * bindings/python/codegen/definitions.py:
14678 * bindings/python/codegen/defsparser.py:
14679 * bindings/python/codegen/docextract.py:
14680 * bindings/python/codegen/docgen.py:
14681 * bindings/python/codegen/fileprefix.override:
14682 * bindings/python/codegen/fileprefixmodule.c:
14683 * bindings/python/codegen/h2def.py:
14684 * bindings/python/codegen/mergedefs.py:
14685 * bindings/python/codegen/mkskel.py:
14686 * bindings/python/codegen/override.py:
14687 * bindings/python/codegen/reversewrapper.py:
14688 * bindings/python/codegen/scmexpr.py:
14689 * bindings/python/rtspserver-types.defs:
14690 * bindings/python/rtspserver.defs:
14691 * bindings/python/rtspserver.override:
14692 * bindings/python/rtspservermodule.c:
14694 Add python bindings.
14696 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14698 * bindings/Makefile.am:
14700 Don't go into python dir when requirements for python bindings are missing
14702 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14704 * bindings/Makefile.am:
14705 * bindings/vala/Makefile.am:
14707 Install Vala bindings if vala is available
14709 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14711 * bindings/vala/gst-rtsp-server-0.10.deps:
14712 * bindings/vala/gst-rtsp-server-0.10.vapi:
14713 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14714 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14715 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14716 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14717 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14718 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14719 Regenerated Vala bindings
14721 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14723 * bindings/vala/gst-rtsp-server.vapi:
14724 * bindings/vala/packages/gst-rtsp-server.metadata:
14725 Fixed typo in included headers for vala bindings
14727 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14731 * pkgconfig/Makefile.am:
14732 * pkgconfig/gst-rtsp-server.pc.in:
14733 Added pkgconfig file
14735 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14737 * bindings/vala/gst-rtsp-server.vapi:
14738 * bindings/vala/packages/gst-rtsp-server.excludes:
14739 * bindings/vala/packages/gst-rtsp-server.gi:
14740 * bindings/vala/packages/gst-rtsp-server.metadata:
14741 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14743 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14745 * bindings/vala/gst-rtsp-server.vapi:
14746 * bindings/vala/packages/gst-rtsp-server.deps:
14747 * bindings/vala/packages/gst-rtsp-server.files:
14748 * bindings/vala/packages/gst-rtsp-server.gi:
14749 * bindings/vala/packages/gst-rtsp-server.metadata:
14750 * bindings/vala/packages/gst-rtsp-server.namespace:
14751 Added Vala bindings
14753 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14755 * gst/rtsp-server/rtsp-session.c:
14756 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14758 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14760 * examples/Makefile.am:
14761 * gst/rtsp-server/Makefile.am:
14762 Put GStreamer version in library name
14764 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14766 * examples/Makefile.am:
14767 * gst/rtsp-server/Makefile.am:
14768 Fix some issues to pass distcheck
14770 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14772 * gst/rtsp-server/rtsp-server.c:
14773 Added port property to GstRTSPServer class.
14775 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14780 * examples/Makefile.am:
14783 * gst/rtsp-server/Makefile.am:
14784 * gst/rtsp-server/rtsp-client.c:
14785 * gst/rtsp-server/rtsp-client.h:
14786 * gst/rtsp-server/rtsp-media.c:
14787 * gst/rtsp-server/rtsp-media.h:
14788 * gst/rtsp-server/rtsp-server.c:
14789 * gst/rtsp-server/rtsp-server.h:
14790 * gst/rtsp-server/rtsp-session-pool.c:
14791 * gst/rtsp-server/rtsp-session-pool.h:
14792 * gst/rtsp-server/rtsp-session.c:
14793 * gst/rtsp-server/rtsp-session.h:
14795 Split in library and example program
14797 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14799 * src/rtsp-client.h:
14800 Removed obsolete variable
14802 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14804 * src/rtsp-client.c:
14805 * src/rtsp-client.h:
14806 Removed pipeline variable GstRTSPClient, because it's only used in one function
14808 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14810 * src/rtsp-media.c:
14811 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14813 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14815 * src/rtsp-session.c:
14816 Initialize some more vars.
14818 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14820 * src/rtsp-session.c:
14821 Initialize variable to avoid compiler warning.
14823 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14826 Add a reasonable generic .gitignore