3 2022-10-04 01:14:01 +0100 Tim-Philipp Müller <tim@centricular.com>
7 * docs/plugins/gst_plugins_cache.json:
8 * gst-rtsp-server.doap:
12 2022-10-04 01:13:59 +0100 Tim-Philipp Müller <tim@centricular.com>
15 Update ChangeLogs for 1.21.1
17 2022-09-21 19:19:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
20 meson: Use implicit builtin dirs in pkgconfig generation
21 Starting with Meson 0.62, meson automatically populates the variables
22 list in the pkgconfig file if you reference builtin directories in the
23 pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
24 We need this, because ${prefix}/libexec is a hard-coded value which is
25 incorrect on, for example, Debian.
26 Bump requirement to 0.62, and remove version compares that retained
27 support for older Meson versions.
28 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
29 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
31 2021-03-24 14:20:18 -0500 Zebediah Figura <z.figura12@gmail.com>
34 meson: Build with -Wl,-z,nodelete to prevent unloading of dynamic libraries and plugins
35 GLib made the unfortunate decision to prevent libgobject from ever being
36 unloaded, which means that now any library which registers a static type
37 can't ever be unloaded either (and any library that depends on those,
39 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
41 2022-09-05 13:28:18 +1200 Chris Wiggins <chris@chriswiggins.co.nz>
43 * gst/rtsp-server/rtsp-context.c:
44 * gst/rtsp-server/rtsp-context.h:
45 rtsp-server: context: Add method to set the RTSPToken on some RTSPContext
47 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2979>
49 2022-08-24 19:50:19 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
51 * gst/rtsp-server/rtsp-server-internal.h:
52 * gst/rtsp-server/rtsp-stream-transport.c:
53 * gst/rtsp-server/rtsp-stream.c:
54 gst-rtsp-server: Fix pushing backlog to client
55 Check back pressure of a stream transport before popping buffer from its backlog.
56 If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
58 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
60 2022-09-02 16:31:54 +0300 Sebastian Dröge <sebastian@centricular.com>
62 * gst/rtsp-server/rtsp-stream.c:
63 rtsp-server: stream: Don't loop forever if binding to the multicast address fails
64 The address/port is pre-defined by the caller of the function, so
65 retrying is only going to loop forever.
66 Ideally the multicast address should be checked after allocating but
67 this doesn't happen currently, so it's better to error out cleanly then
68 to loop forever trying the same address.
69 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
71 2022-09-01 15:11:31 -0400 Thibault Saunier <tsaunier@igalia.com>
73 * gst/rtsp-sink/meson.build:
75 meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
76 Removing some copy pasted code
77 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
79 2022-09-01 11:51:48 -0400 Thibault Saunier <tsaunier@igalia.com>
82 * gst/rtsp-server/meson.build:
84 meson: Namespace the plugins_doc_dep/libraries variables
85 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
87 2022-08-31 18:44:14 -0400 Thibault Saunier <tsaunier@igalia.com>
90 meson: Rename plugins list and make them "dependency" objects
91 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
93 2022-05-25 18:40:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
95 * gst/rtsp-sink/gstrtspclientsink.c:
96 rtsp+rtmp: Forward warning added to tls-validation-flags to our users
97 With the 2.72 release, glib-networking developers have decided that
98 TLS certificate validation cannot be implemented correctly by them, so
99 they've deprecated it.
100 In a nutshell: a cert can have several validation errors, but there
101 are no guarantees that the TLS backend will return all those errors,
102 and things are made even more complicated by the fact that the list of
103 errors might refer to certs that are added for backwards-compat and
104 won't actually be used by the TLS library.
105 Our best option is to ignore the deprecation and pass the warning onto
106 users so they can make an appropriate security decision regarding
108 We can't deprecate the tls-validation-flags property because it is
109 very useful when connecting to RTSP cameras that will never get
110 updates to fix certificate errors.
111 Relevant upstream merge requests / issues:
112 https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214
113 https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179
114 https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
115 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
117 2022-07-12 16:58:00 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
119 * gst/rtsp-server/rtsp-client.c:
120 rtsp-client: Fix url for generating key in media factory
121 The mount point at / can be accessed by both the URL forms rtsp://<IP>:<PORT> and rtsp://<IP>:<PORT>/.
122 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.
123 This commit creates a new GstRTSPUrl as the normalized url to send to gst_rtsp_media_factory_construct().
124 Fixes:https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1297
125 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2681>
127 2022-06-29 10:55:13 +0100 Tim-Philipp Müller <tim@centricular.com>
130 coding style: allow declarations after statement
131 See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
132 and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
133 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2683>
135 2022-06-14 16:18:35 +0100 Tim-Philipp Müller <tim@centricular.com>
138 * docs/plugins/gst_plugins_cache.json:
139 * docs/plugins/index.md:
140 * docs/plugins/sitemap.txt:
141 docs: make sure rtspclientsink plugin docs index page is called index.html
142 .. instead of plugin-index.html.
143 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2592>
145 2022-04-06 12:56:30 +0100 Tim-Philipp Müller <tim@centricular.com>
148 Bump GLib requirement to >= 2.62
149 Can't require 2.64 yet because of
150 https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323
151 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
153 2022-05-16 18:06:16 +0200 Patricia Muscalu <patricia@axis.com>
155 * gst/rtsp-server/rtsp-media.c:
156 rtsp-media: Correct logic on GstRTSPStreamBlocking message reception
157 We must take into account the receiving streams as well when calculating
158 the expected number of the received GstRTSPStreamBlocking messages.
159 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2429>
161 2022-04-27 01:13:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
163 * tests/check/gst/onvif.c:
164 tests/onvif: improve robustness
165 The previous iteration of the code was inferring the type of the
166 frame by looking at the overall size of the gst-payloaded packet.
167 It is more robust to actually parse the payload and look at the
168 actual data buffers it contains.
169 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
171 2022-04-27 01:10:46 +0200 Mathieu Duponchelle <mathieu@centricular.com>
173 * tests/check/gst/onvif.c:
174 tests/onvif: don't push buffers outside segment
175 segment->stop is exclusive, so in reverse playback mode we do not
176 need to output a buffer at that position as it will simply get
178 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
180 2022-02-15 13:39:43 +0000 Pierre Bourré <pierre.moltess@gmail.com>
182 * gst/rtsp-sink/gstrtspclientsink.c:
183 rtspclientsink: fix possible shutdown deadlock collect_streams()
184 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1696>
186 2022-04-07 19:14:27 +0300 Sebastian Dröge <sebastian@centricular.com>
188 * gst/rtsp-server/rtsp-sdp.c:
189 rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME
190 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
192 2022-04-13 14:34:57 +0200 Marc Leeman <m.leeman@televic.com>
194 * gst/rtsp-server/rtsp-stream.c:
195 gst-rtsp-server: minor spelling fixes
196 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>
198 2022-03-25 15:00:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
200 * examples/meson.build:
202 Remove glib and gobject dependencies everywhere
203 They are part of gst_dep already and we have to make sure to always have
204 gst_dep. The order in dependencies matters, because it is also the order
205 in which Meson will set -I args. We want gstreamer's config.h to take
206 precedence over glib's private config.h when it's a subproject.
207 While at it, remove useless fallback args for gmodule/gio dependencies,
208 only gstreamer core needs it.
209 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
211 2022-03-28 21:03:16 +1100 Matthew Waters <matthew@centricular.com>
213 * gst/rtsp-server/rtsp-stream.c:
214 rtsp-stream: remove unused variable:
216 ../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
217 guint n_messages = 0;
219 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
221 2022-03-18 13:42:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
224 meson: Bump all meson requirements to 0.60
225 Lots of new warnings ever since
226 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1934
227 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
229 2022-02-23 17:39:18 +0100 Vivienne Watermeier <vwatermeier@igalia.com>
231 * gst/rtsp-server/rtsp-token.c:
232 documentation: improve misleading wording
233 The documentation for several gst_*_writable_structure functions stated
234 that they would never return NULL, without making clear that the passed
235 object is required to be writable. This changes the wording in those
236 cases to make that requirement more clear.
237 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
239 2022-02-10 08:01:02 +0100 Branko Subasic <branko@axis.com>
241 * examples/test-onvif-server.c:
242 * tests/check/gst/onvif.c:
243 rtponviftimestamp: add support for using reference timestamps
244 Make it posible to configure the element to obtain the timestamps from
245 reference timestamp meta data instead of using the ntp-offset property,
246 or estimating its own offset. Currently the only time format supported
247 is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.
248 In addition the custom event GstNtpOffset has been renamed to
249 GstOnvifTimestamp, to reflect that it is not necessarily used to convey
250 the ntp-offset. As a consequence we had to modify a couple of files in
251 the rtsp-server as well.
253 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
255 2022-02-18 16:05:53 +0100 Branko Subasic <branko@axis.com>
257 * tests/check/gst/onvif.c:
258 * tests/check/gst/rtspserver.c:
259 * tests/check/gst/stream.c:
260 gst-rtsp-server: Plug a few memory leaks in tests
261 Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and
262 gst_stream tests by running the tests in valgrind.
263 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
265 2022-03-07 09:14:46 +0100 Branko Subasic <branko@axis.com>
267 * gst/rtsp-server/rtsp-client.c:
268 gst-rtsp-server: fix race in rtsp-client
269 When tunneling over HTTP, if connection on the second channel happens
270 before the control timer is created we may trigger an assert in
271 rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
272 attaching the client thread to the context.
274 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
276 2022-02-04 11:15:47 +0000 Tim-Philipp Müller <tim@centricular.com>
278 * docs/gst_plugins_cache.json:
281 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
283 === release 1.20.0 ===
285 2022-02-03 19:53:25 +0000 Tim-Philipp Müller <tim@centricular.com>
290 * docs/gst_plugins_cache.json:
291 * gst-rtsp-server.doap:
295 2022-02-03 19:53:18 +0000 Tim-Philipp Müller <tim@centricular.com>
298 Update ChangeLogs for 1.20.0
300 === release 1.19.90 ===
302 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
307 * docs/gst_plugins_cache.json:
308 * gst-rtsp-server.doap:
312 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
315 Update ChangeLogs for 1.19.90
317 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
319 * examples/test-appsrc2.c:
320 gst-rtsp-server: Fix leak in appsrc2 example
321 In the need-data appsrc callback, a buffer is pulled from the
322 appsink. This buffer is then copied so that metadata is writable.
323 The copy is pushed to the appsrc but it doesn't take ownership
324 of the buffer so we need to manually unref it. The original buffer
325 is finally unreffed when the sample is freed.
326 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
328 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
332 meson: Add explicit check: kwarg to all run_command() calls
333 This is required since Meson 0.61.0, and causes a warning to be
335 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
336 https://github.com/mesonbuild/meson/issues/9300
337 This exposed a bunch of places where we had broken run_command()
338 calls, unnecessary run_command() calls, and places where check: true
340 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
342 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
344 * gst/rtsp-server/meson.build:
345 rtsp-server: add gst_dep to gst_rtsp_server_deps
346 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
347 will avoid the following build failure, because the correct girdir
348 location will be retrieved from gstreamer-1.0.pc:
349 /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
350 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
351 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
352 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
353 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
355 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
356 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
358 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
360 * gst/rtsp-server/rtsp-stream.c:
361 rtsp-stream: fix get_rates raciness
362 Prior to this patch, we considered that a stream was blocking
363 whenever a pad probe was triggered for either the RTP pad or
365 This led to situations where we subsequently unblocked and expected
366 to find a segment on the RTP pad, which was racy.
367 Instead, we now only consider that the stream is blocking when
368 the pad probe for the RTP pad has triggered with a blockable object
369 (buffer, buffer list, gap event).
370 The RTCP pad is simply blocked without affecting the state of the
373 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
375 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
377 * docs/gst_plugins_cache.json:
381 === release 1.19.3 ===
383 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
388 * docs/gst_plugins_cache.json:
389 * gst-rtsp-server.doap:
393 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
396 Update ChangeLogs for 1.19.3
398 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
401 meson: require matching GStreamer dep versions for unstable development releases
402 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
403 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
405 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
407 * tests/check/meson.build:
408 meson: update for meson.build_root() and .build_source() deprecation
409 -> use meson.project_build_root() or .global_build_root() instead.
410 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
412 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
415 * tests/check/meson.build:
416 meson: update for dep.get_pkgconfig_variable() deprecation
417 ... in favour of dep.get_variable('foo', ..) which in some
418 cases allows for further cleanups in future since we can
419 extract variables from pkg-config dependencies as well as
420 internal dependencies using this mechanism.
421 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
423 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
425 * gst/rtsp-server/meson.build:
426 * gst/rtsp-sink/meson.build:
427 rtsp-server: define G_LOG_DOMAIN
429 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
431 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
434 meson: bump meson requirement to >= 0.59
435 For monorepo build and ugly/bad, for advanced feature
436 option API like get_option('xyz').required(..) which
437 we use in combination with the 'gpl' option.
438 For rest of modules for consistency (people will likely
439 use newer features based on the top-level requirement).
440 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
442 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
445 meson: Streamline the way we detect when to build documentation
446 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
448 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
451 * gst/rtsp-server/meson.build:
453 meson: List libraries and their corresponding gir definition
454 Introduces a `libraries` variable that contains all libraries in a
455 list with the following format:
459 'lib': library_object
460 'gir': [ {full gir definition in a dict } ]
465 It therefore refactors the way we build the gir so that we can reuse the
466 same information to build them against 'gstreamer-full' in gst-build
467 when linking statically
468 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
470 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
472 * gst/rtsp-server/meson.build:
473 meson: Mark files as files()
474 Making it more robust and future proof
475 And fix issues that it creates
476 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
478 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
480 * gst/rtsp-server/rtsp-media.c:
481 rtsp-media: Unprepare suspended medias too
482 Previously suspended medias immediately reached the UNPREPARED state
483 without going through the media's unprepare() vfunc. This didn't allow
484 the media subclass to do any additional cleanup, and for example the
485 shutdown-eos property of GstRTSPMedia was ignored.
486 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
488 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
490 * gst/rtsp-server/rtsp-media.c:
491 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
492 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
494 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
496 * gst/rtsp-server/rtsp-client.c:
497 * gst/rtsp-server/rtsp-session.c:
498 * gst/rtsp-server/rtsp-session.h:
499 rtsp-client: make sure sessmedia will not get freed while used
500 handle_*_request() functions were all retrieving the session media from
501 the session by calling gst_rtsp_session_get_media () which is a transfer-none
502 call. If a session timeout happens at that time, the session media may get freed
503 making the pointer invalid..
505 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
507 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
509 * gst/rtsp-server/rtsp-media.c:
510 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
511 Previously the status was only changed for other medias.
512 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
514 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
516 * gst/rtsp-server/rtsp-session.c:
517 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
518 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
519 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
521 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
524 doc: update IRC links to OFTC
525 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
527 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
529 * docs/gst_plugins_cache.json:
532 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
534 === release 1.19.2 ===
536 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
541 * docs/gst_plugins_cache.json:
542 * gst-rtsp-server.doap:
546 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
548 * gst/rtsp-server/rtsp-media.c:
549 * gst/rtsp-server/rtsp-stream.c:
550 * gst/rtsp-server/rtsp-stream.h:
551 * gst/rtsp-sink/gstrtspclientsink.c:
552 Protection against early RTCP packets.
553 When receiving RTCP packets early the funnel is not ready yet and
554 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
555 This causes the thread that handle RTCP packets to go to pause mode.
556 Since this thread is in pause mode there will be no further callbacks to
557 handle keep-alive for incoming RTCP packets. This will make the session
558 time out if the client is not using another keep-alive mechanism.
559 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
560 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
562 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
566 Update COPYING.LIB, COPYING files
567 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
569 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
571 * docs/gst_plugins_cache.json:
575 === release 1.19.1 ===
577 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
582 * docs/gst_plugins_cache.json:
583 * gst-rtsp-server.doap:
587 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
589 * gst/rtsp-server/rtsp-stream.c:
590 rtsp-stream: use new gst_buffer_new_memdup()
591 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
593 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
595 * gst/rtsp-server/rtsp-media-factory-uri.c:
596 rtsp-media: fix leak when adding converter
597 Free the previous caps before reusing the variable for the converter caps.
598 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
600 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
602 * gst/rtsp-server/rtsp-client.c:
603 rtsp-client: fix leak adding headers
604 gst_rtsp_message_add_header() makes a copy of the header, instead
606 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
608 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
610 * gst/rtsp-server/rtsp-stream.c:
611 Use gst_element_request_pad_simple...
612 Instead of the deprecated gst_element_get_request_pad.
613 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
615 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
617 * gst/rtsp-server/rtsp-media.c:
618 rtsp-media: Ensure the bus watch is removed during unprepare
619 It's possible for the destruction of the source to be delayed.
620 Instead of relying on the dispose() to remove the bus watch, do
622 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
624 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
627 docs: minor spelling correction in README
628 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
630 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
632 * examples/test-replay-server.c:
633 test-replay-server: minor spelling corrections
634 Bumped on these while investigating the example code.
635 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
637 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
639 * tests/check/gst/stream.c:
640 tests: Don't fail tests if IPv6 not available.
641 On computers with IPv6 disabled it shouldn't result in a test failure.
642 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
644 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
646 * gst/rtsp-server/rtsp-media.c:
647 rtsp-media: Add one more case to seek avoidance
648 This is an extension to the previous commit. There can also be cases where the
649 start position is not specified, in those cases we should also avoid doing
650 seeking unless it's forced.
651 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
653 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
655 * gst/rtsp-server/rtsp-media.c:
656 rtsp-media: Improve skipping trickmode seek.
657 We can also skip the seek if the end range is already
659 Avoids initial seek on play start if playing full stream.
660 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
662 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
664 * gst/rtsp-sink/gstrtspclientsink.c:
665 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
666 It's sufficient to run them during the FIRST stage instead of in both.
667 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
669 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
671 * tests/check/gst/rtspclientsink.c:
672 tests: rtspclientsink: fix some leaks
673 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
675 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
677 * gst/rtsp-sink/gstrtspclientsink.c:
678 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
679 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
681 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
683 * tests/check/gst/rtspclientsink.c:
684 rtspclientsink: add unit test for potential shutdown deadlock
685 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
687 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
689 * gst/rtsp-sink/gstrtspclientsink.c:
690 rtspclientsink: fix deadlock on shutdown before preroll
691 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
692 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
694 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
696 * gst/rtsp-server/rtsp-stream.c:
697 rtsp-stream: avoid deadlock in send_func
698 Currently the send_func() runs in a thread of its own which is started
699 the first time we enter handle_new_sample(). It runs in an outer loop
700 until priv->continue_sending is FALSE, which happens when a TEARDOWN
701 request is received. We use a local variable, cont, which is initialized
702 to TRUE, meaning that we will always enter the outer loop, and at the
703 end of the outer loop we assign it the value of priv->continue_sending.
704 Within the outer loop there is an inner loop, where we wait to be
705 signaled when there is more data to send. The inner loop is exited when
706 priv->send_cookie has changed value, which it does when more data is
707 available or when a TEARDOWN has been received.
708 But if we get a TEARDOWN before send_func() is entered we will get stuck
709 in the inner loop because no one will increase priv->session_cookie
711 By not entering the outer loop in send_func() if priv->continue_sending
712 is FALSE we make sure that we do not get stuck in send_func()'s inner
713 loop should we receive a TEARDOWN before the send thread has started.
714 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
715 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
717 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
719 * gst/rtsp-server/rtsp-client.c:
720 rtsp-client: cleanup transports during TEARDOWN
721 When tunneling RTP over RTSP the stream transports are stored in a hash
722 table in the GstRTSPClientPrivate struct. They are used for, among other
723 things, mapping channel id to stream transports when receiving data from
724 the client. The stream tranports are created and added to the hash table
725 in handle_setup_request(), but unfortuately they are not removed in
726 handle_teardown_request(). This means that if the client sends data on
727 the RTSP connection after it has sent the TEARDOWN, which is often the
728 case when audio backchannel is enabled, handle_data() will still be able
729 to map the channel to a session transport and pass the data along to it.
730 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
731 because the stream is no longer joined to a bin.
732 We avoid this by removing the stream transports from the hash table when
733 we handle the TEARDOWN request.
734 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
736 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
738 * docs/gst_plugins_cache.json:
739 * gst/rtsp-sink/gstrtspclientsink.c:
740 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
741 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
743 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
745 * tests/check/gst/client.c:
746 Add test cases for mountpoint of '/'
747 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
749 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
751 * gst/rtsp-server/rtsp-client.c:
752 * gst/rtsp-server/rtsp-mount-points.c:
753 * gst/rtsp-server/rtsp-session-media.c:
754 Make a mount point of "/" work correctly.
755 As far as I can tell, this is neither explicitly allowed nor
756 forbidden by RFC 7826.
757 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
758 use in the wild (presumably with non-GStreamer servers).
759 GStreamer's prior behavior was confusing, in that
760 gst_rtsp_mount_points_add_factory() would appear to accept a mount
761 path of "" or "/", but later connection attempts would fail with a
762 "media not found" error.
763 This commit makes a mount path of "/" work for either form of URL,
764 while an empty mount path ("") is rejected and logs a warning.
765 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
767 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
769 * docs/gst_plugins_cache.json:
770 * gst/rtsp-sink/gstrtspclientsink.c:
771 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
772 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
774 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
776 * gst/rtsp-server/rtsp-media.c:
777 rtsp-media: Only count senders when counting blocked streams
778 Only sender streams sends the GstRTSPStreamBlocking message, so only
779 these should be counted before setting media status to prepared.
780 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
782 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
784 * gst/rtsp-sink/gstrtspclientsink.c:
785 rtspclientsink add proper support for uri queries
786 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
788 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
790 * gst/rtsp-server/rtsp-client.c:
791 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
792 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
793 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
795 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
797 * gst/rtsp-server/rtsp-stream.c:
798 rtsp-stream: collect a clock_rate when blocking
799 This lets us provide a clock_rate in a fashion similar to the
800 other code paths in get_rtpinfo()
801 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
803 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
805 * gst/rtsp-server/rtsp-media.c:
806 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
807 Otherwise this will cause memory corruption as the property expects a 64
809 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
811 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
813 * gst/rtsp-server/rtsp-media.c:
814 * gst/rtsp-server/rtsp-stream.c:
815 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
816 To prevent cases with prerolling when the inactive stream prerolls first
817 and the server proceeds without waiting for the active stream, we will
818 ignore GstRTSPStreamBlocking messages from incomplete streams. When
819 there are no complete streams (during DESCRIBE), we will listen to all
821 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
823 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
825 * tests/check/gst/media.c:
826 * tests/check/meson.build:
827 * tests/files/test.avi:
828 media test: Add test for seeking one active stream with a demuxer
829 Add another seek_one_active_stream test but with a demuxer. The demuxer
830 will flush both streams in opposed to the existing test which only
831 flushes the active stream. This will help exposing problems with the
832 prerolling process after a flushing seek.
833 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
835 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
837 * gst/rtsp-server/meson.build:
839 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
840 * pkgconfig/gstreamer-rtsp-server.pc.in:
841 * pkgconfig/meson.build:
842 Meson: Use pkg-config generator
843 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
845 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
848 meson: update glib minimum version to 2.56
849 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
851 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
853 * examples/test-launch.c:
854 * gst/rtsp-server/rtsp-media-factory.c:
855 * gst/rtsp-server/rtsp-media-factory.h:
856 * gst/rtsp-server/rtsp-media.c:
857 * gst/rtsp-server/rtsp-server-internal.h:
858 * gst/rtsp-server/rtsp-stream.c:
859 * tests/check/gst/client.c:
860 rtsp-media-factory: expose API to disable RTCP
861 This is supported by the RFC, and can be useful on systems where
862 allocating two consecutive ports is problematic, and RTCP is not
864 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
866 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
868 * hooks/pre-commit.hook:
870 git: use our standard pre commit hook
871 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
873 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
875 * gst/rtsp-server/rtsp-stream.c:
876 rtsp-stream: make use of blocked_running_time in query_position
877 When blocking, the sink element will not have received a buffer
878 yet and the position query will fail. Instead, we make use of
879 the running time of the buffer we blocked on.
880 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
882 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
884 * gst/rtsp-server/rtsp-stream.c:
885 rtsp-stream: collect rtp info when blocking
886 We don't unblock the stream anymore before replying to the
887 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
888 so the sinks don't have a last-sample after potentially flush
889 seeking. seek_trickmode waits for preroll however, which means
890 the stream will block and wait for a first buffer. Subsequent
891 calls to get_rtpinfo() can thus make use of the information.
892 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
893 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
895 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
897 * examples/meson.build:
898 * examples/test-replay-server.c:
899 * examples/test-replay-server.h:
900 examples: Add an example for loop playback
901 This demo example shows a way of file loop playback of a given source.
902 Note that client seek request is not properly implemented yet.
903 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
905 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
907 * gst/rtsp-server/rtsp-media.c:
908 rtsp-media: Plug memory leak
909 The get-storage signal of rtpbin increases the ref count of the storage.
910 So we have to unref it after usage.
911 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
913 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
915 * gst/rtsp-server/rtsp-media.c:
916 rtsp-media: Get rates only on sender streams
917 When play a media with both sender and receiver stream, like ONVIF
918 back channel audio in, gst_rtsp_media_get_rates call
919 gst_rtsp_stream_get_rates for each stream to set the rates. But
920 gst_rtsp_stream_get_rates return false for the receiver steam, which
921 lead a g_assert crash.
922 Instead to get rates on all streams, now just get rates on sender
924 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
926 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
928 * gst/rtsp-server/rtsp-media.c:
929 * gst/rtsp-server/rtsp-server-internal.h:
930 * gst/rtsp-server/rtsp-stream.c:
931 rtsp-media: set a 0 storage size for TCP receivers
932 ulpfec correction is obviously useless when receiving a stream
933 over TCP, and in TCP modes the rtp storage receives non
934 timestamped buffers, causing it to queue buffers indefinitely,
935 until the queue grows so large that sanity checks kick in and
936 warnings start to get emitted.
937 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
939 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
941 * gst/rtsp-server/rtsp-stream.c:
942 rtsp-stream: preroll on gap events
943 This allows negotiating a SDP with all streams present, but only
944 start sending packets at some later point in time
945 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
947 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
949 * gst/rtsp-server/rtsp-media.c:
950 rtsp-media: do not unblock on unsuspend
951 rtsp_media_unsuspend() is called from handle_play_request()
952 before sending the play response. Unblocking the streams here
953 was causing data to be sent out before the client was ready
954 to handle it, with obvious side effects such as initial packets
955 getting discarded, causing decoding errors.
956 Instead we can simply let the media streams be unblocked when
957 the state of the media is set to PLAYING, which occurs after
958 sending the play response.
959 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
961 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
964 ci: include template from gst-ci master branch again
966 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
968 * docs/gst_plugins_cache.json:
972 === release 1.18.0 ===
974 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
980 * docs/gst_plugins_cache.json:
981 * gst-rtsp-server.doap:
985 === release 1.17.90 ===
987 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
992 * docs/gst_plugins_cache.json:
993 * gst-rtsp-server.doap:
997 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
999 * gst/rtsp-server/rtsp-thread-pool.c:
1000 rtsp-thread-pool.c: fix clang 10 warning
1001 clang 10 is complaining about incompatible types due to the
1004 ../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]
1006 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1008 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
1010 * gst/rtsp-server/rtsp-thread-pool.c:
1011 rtsp-thread-pool.c: fix clang 10 warning
1012 clang 10 is complaining about incompatible types due to the
1015 ../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]
1017 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1019 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
1021 * gst/rtsp-server/rtsp-sdp.c:
1022 rtsp-sdp: Fix resource leak in mikey messsage
1023 Fixed a resource leak for mikey message while adding crypto session
1025 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
1027 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
1030 * scripts/extract-release-date-from-doap-file.py:
1031 meson: set release date from .doap file for releases
1032 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
1034 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1036 * gst/rtsp-server/rtsp-stream.c:
1037 rtsp-stream: explicitly set caps on udpsrc elements
1038 This causes them to send caps events before data flow, which is
1039 usually a pretty correct thing to do!
1040 Not doing so manifested in a bug where ssrcdemux wouldn't forward
1041 the caps it had received with an extra ssrc field, as it hadn't
1042 received any caps event.
1044 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
1046 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1048 * docs/gst_plugins_cache.json:
1052 === release 1.17.2 ===
1054 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1059 * docs/gst_plugins_cache.json:
1060 * gst-rtsp-server.doap:
1064 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
1066 * docs/gst_plugins_cache.json:
1067 doc: Stop documenting properties from parents
1069 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1071 * docs/gst_plugins_cache.json:
1072 docs: Fix version in the plugins cache
1073 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1075 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1077 * gst/rtsp-sink/gstrtspclientsink.c:
1078 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
1079 It's deprecated, unneeded and doesn't do anything anymore.
1080 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1082 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
1087 === release 1.17.1 ===
1089 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
1094 * docs/gst_plugins_cache.json:
1095 * gst-rtsp-server.doap:
1099 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1101 * gst/rtsp-server/rtsp-media.c:
1102 rtsp-media: Add/configure transports when completing the pipeline
1103 Otherwise the transports are not set up yet during the PLAY request
1104 handling when unsuspending (and thus unblocking) the media.
1105 In case of live pipelines this then causes the first few packets to go
1106 to the sinks before they know what to do with them, and they simply
1107 discard them which is rather suboptimal in case of keyframes.
1108 For non-live pipelines this is not a problem because the sink will still
1109 be PAUSED and as such not send out the data yet but wait until it goes
1110 to PLAYING, which is late enough.
1111 Adding the transports multiple times is not a problem: if the transport
1112 is already added it won't be added another time and TRUE will be
1114 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
1116 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
1117 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1119 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1121 * gst/rtsp-server/rtsp-media.c:
1122 rtsp-media: Fix misleading comment
1123 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1125 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
1127 * gst/rtsp-server/rtsp-media.c:
1128 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
1129 The pad probes are not needed anymore at this point and later when
1130 reaching buffering 100% only the state is changed, no unblocking
1132 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1134 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
1136 * gst/rtsp-server/rtsp-media.c:
1137 rtsp-media: Remove duplicated media_unblock() function
1138 It does literally the same as media_streams_set_blocked(FALSE).
1139 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1141 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
1143 * examples/test-onvif-server.c:
1144 test-onvif-server: cast ntp-offset property value to 64 bit
1145 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
1147 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
1149 * docs/gst_plugins_cache.json:
1150 docs: Update plugins cache
1152 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1154 * examples/test-onvif-server.c:
1155 * examples/test-onvif-server.h:
1156 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1157 onvif-media-factory: define autoptr cleanup function
1158 And have the factory in the onvif-server example inherit from
1159 GstRTSPOnvifMediaFactory.
1160 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
1162 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
1164 * docs/gst_plugins_cache.json:
1165 docs: Update plugins cache
1167 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
1169 * tests/check/gst/rtspserver.c:
1170 tests: enforce I420 format
1171 Test was not enforcing a video format on videotestsrc. I420 was picked as it
1172 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
1173 true (gst-plugins-base!689).
1174 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
1176 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1178 * gst/rtsp-sink/gstrtspclientsink.c:
1179 plugins: uddate gst_type_mark_as_plugin_api() calls
1181 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
1184 doc: Require hotdoc >= 0.11.0
1186 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1188 * docs/gst_plugins_cache.json:
1189 docs: Update gst_plugins_cache.json
1191 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1193 * gst/rtsp-sink/gstrtspclientsink.c:
1194 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
1196 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1198 * gst/rtsp-server/meson.build:
1199 meson: gir: remove bogus sources_top_dir kwarg
1200 Doesn't actually exist. Was fixed differently in Meson
1201 so that the user doesn't have to specify it.
1202 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
1204 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1206 * tests/check/meson.build:
1207 tests: put registry into tests/check not the gst/ subdir
1208 Underscorify the test name before setting GST_REGISTRY,
1209 so the registry actually ends up in the current build dir
1210 and not some subdir.
1211 For consistency with the other modules, but should also
1212 avoid problems on windows.
1213 Also fix indentation of environment block.
1214 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1216 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
1218 * tests/check/meson.build:
1219 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
1220 If core is built as a subproject (e.g. as in gst-build), make sure to use
1221 the gst-plugin-scanner from the built subproject. Without this, gstreamer
1222 might accidentally use the gst-plugin-scanner from the install prefix if
1223 that exists, which in turn might drag in gst library versions we didn't
1224 mean to drag in. Those gst library versions might then be older than
1225 what our current build needs, and might cause our newly-built plugins
1226 to get blacklisted in the test registry because they rely on a symbol
1227 that the wrongly-pulled in gst lib doesn't have.
1228 This should fix running of unit tests in gst-build when invoking
1229 meson test or ninja test from outside the devenv for the case where
1230 there is an older or different-version gst-plugin-scanner installed
1231 in the install prefix.
1232 In case no gst-plugin-scanner is installed in the install prefix, this
1233 will fix "GStreamer-WARNING: External plugin loader failed. This most
1234 likely means that the plugin loader helper binary was not found or
1235 could not be run. You might need to set the GST_PLUGIN_SCANNER
1236 environment variable if your setup is unusual." warnings when running
1238 In the case where we find GStreamer core via pkg-config we use
1239 a newly-added pkg-config var "pluginscannerdir" to get the right
1240 directory. This has the benefit of working transparently for both
1241 installed and uninstalled pkg-config files/setups.
1242 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1244 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
1246 * tests/check/meson.build:
1247 tests: gst-plugins-base and -bad plugins are required for the unit tests
1248 Make hard requirement until we have more fine-grained control
1249 in the unit tests. Of course the presence of the .pc file doesn't
1250 imply that the plugins we need are actually there, but it's at
1251 least a step in the right direction.
1252 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1254 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1256 * tests/check/meson.build:
1257 tests: pick up rtsp-server plugins from build directory only
1258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1260 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
1262 * gst/rtsp-server/rtsp-media.c:
1263 rtsp-media: wait for all GstRTSPStreamBlocking messages
1264 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
1265 each active stream when checking if all streams are blocked.
1266 Without this change there will be a race condition when using two or
1267 more streams and rtsp-media receives a GstRTSPStreamBlocking message
1268 from one of the streams. This is because rtsp-media then checks if all
1269 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
1270 stream. This function call returns TRUE if the stream has sent a
1271 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
1272 receive this message. This would then result in that rtsp-media
1273 erroneously thinks it is blocking all streams which could result in
1274 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
1275 preroll, this could result in that rtsp-media thinks that the pipeline
1276 is prerolled even though that might not be the case.
1277 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
1279 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
1281 * gst/rtsp-server/rtsp-media.c:
1282 rtsp-media: update expected_async_done during suspend
1283 Set expected_async_done to FALSE in default_suspend() if a state change
1284 occurs and the return value from set_target_state() is something other
1285 than GST_STATE_CHANGE_ASYNC.
1286 Without this change there is a risk that expected_async_done will be
1287 TRUE even though no asynchronous state change is taking place. This
1288 could happen if the pipeline is set to PAUSED using
1289 media_set_pipeline_state_locked(), an asynchronous state change starts
1290 and then the media is suspended (which could result in a state change,
1291 aborting the asynchronous state change). If the media is suspended
1292 before the asynchronous state change ends then expected_async_done will
1293 be TRUE but no asynchronous state change is taking place.
1294 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
1296 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
1298 * gst/rtsp-server/rtsp-client.c:
1299 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1300 There was a race condition where client was being finalized and
1301 concurrently in some other thread the rtsp ctrl timout was relying on
1302 client data that was being freed.
1303 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1304 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1306 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1308 * gst/rtsp-server/rtsp-media-factory.c:
1309 * gst/rtsp-server/rtsp-media-factory.h:
1310 * gst/rtsp-server/rtsp-media.c:
1311 * gst/rtsp-server/rtsp-media.h:
1312 media-factory: complete DSCP QoS setting support
1313 add dscp_qos setting support at factory and media level to setup IP DSCP
1314 field of bounded UDP sinks.
1315 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1316 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1318 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1320 * gst/rtsp-server/rtsp-client.c:
1321 rtsp-client: Fix some race conditions around timeout source removal
1322 We always need to take the lock while accessing it as otherwise another
1323 thread might've removed it in the meantime. Also when destroying and
1324 creating a new one, ensure that the mutex is not shortly unlocked in
1325 between as during that time another one might potentially be created
1327 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1329 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1331 * gst/rtsp-server/rtsp-media.c:
1332 * gst/rtsp-server/rtsp-stream.c:
1333 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1334 And the same for gst_rtsp_stream_get_rates().
1335 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1337 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1339 * examples/test-onvif-server.c:
1340 examples: test-onvif-server: fix compiler warnings on raspbian
1341 Fix printf format for 64-bit variables.
1342 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1344 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1346 * gst/rtsp-server/rtsp-stream-transport.c:
1347 * gst/rtsp-server/rtsp-stream-transport.h:
1348 * gst/rtsp-server/rtsp-stream.c:
1349 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1350 The old API is preserved now and new API was added that provides the
1351 additional parameter to the callback.
1352 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1353 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1355 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1357 * gst/rtsp-server/rtsp-client.c:
1358 rtsp-client: Store the timeout source by pointer instead of id
1359 That way we don't have to retrieve it again from the main context when
1360 destroying it but can directly do so.
1361 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1363 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1365 * gst/rtsp-server/rtsp-client.c:
1366 rtsp-client: Clean up watch/watch context and related state consistently
1367 And assert that it was cleaned up properly before the client is
1368 finalized. If something is still around when the client is shut down
1369 then something went very wrong before.
1370 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1372 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1374 * gst/rtsp-server/rtsp-client.c:
1375 * tests/check/gst/rtspserver.c:
1376 rtsp-client: Combine the pre-session and post-session timeout
1377 They previously used the same state but different mechanisms and
1378 functions, which was difficult to follow, error prone and simply
1380 Also adjust the test for the post-session timeout a bit to be less racy
1381 now that the timing has slightly changed.
1382 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1384 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1386 * gst/rtsp-server/rtsp-client.c:
1387 rtsp-client: Don't ever close the client connection directly when a session is torn down
1388 There might be other sessions that are running over the same RTSP
1389 connection and we should not simply close the client directly if one of
1391 By default the connection will be closed once the client closes it or
1392 the OS does. This behaviour can be adjusted with the
1393 post-session-timeout property, which allows to close it automatically
1394 from the server side after all sessions are gone and the given timeout
1396 This reverts the previous commit.
1397 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1399 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1401 * gst/rtsp-server/rtsp-client.c:
1402 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1403 Instead of closing it never at all. Previously there was only code that
1404 closed the client asynchronously if sending the response happened
1405 asynchrously at a later time.
1406 Thanks to Christian M for debugging this issue.
1407 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1408 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1410 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1412 * gst/rtsp-server/rtsp-stream.c:
1413 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1414 Otherwise no sink is found for multicast sreams and the less accurate
1415 fallback is used to determine the current sequence number and timestamp.
1417 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1419 * gst/rtsp-server/rtsp-auth.c:
1420 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1421 When using the basic authentication scheme, we wouldn't validate that
1422 the authorization field of the credentials is not NULL and pass it on
1423 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1424 dereference the NULL pointer and crash.
1425 A specially crafted (read: invalid) RTSP header can cause this to
1427 As a solution, check for the authorization to be not NULL before
1428 continuing processing it and if it is simply fail authentication.
1429 This fixes CVE-2020-6095 and TALOS-2020-1018.
1430 Discovered by Peter Wang of Cisco ASIG.
1432 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1434 * gst/rtsp-server/rtsp-client.c:
1435 rtsp-client: Use watch_context before unref
1436 Move the usage of priv->watch_context to beginning of function
1437 gst_rtsp_client_finalize. Instead of use it after
1438 g_main_context_unref (priv->watch_context).
1440 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1442 * gst/rtsp-server/rtsp-stream.c:
1443 rtsp-stream: fix deadlock on transport removal
1444 We cannot take the RTSPStream lock while holding a transport backlog
1445 lock, as remove_transport may be called externally, which will
1446 take first the RTSPStream lock then the transport backlog lock.
1448 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1450 * gst/rtsp-server/rtsp-server-internal.h:
1451 * gst/rtsp-server/rtsp-stream-transport.c:
1452 * gst/rtsp-server/rtsp-stream.c:
1453 rtsp-stream: clear backlog when removing transport
1454 This ensures we don't end up calling any of transports' callbacks
1455 with a potentially unreffed user_data (in practice, a client that
1456 may have been removed)
1458 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1460 * gst/rtsp-server/rtsp-stream.c:
1461 rtsp-stream: marshal calls to send_tcp_message to a single thread
1462 In order to address the race condition pointed out at
1463 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1464 we get rid of the send thread pool, and instead spawn and manage
1465 a single thread to pull samples from app sinks and add them to
1466 the transport's backlogs.
1467 Additionally, we now also always go through the backlogs in order
1468 to simplify the logic.
1470 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1472 * gst/rtsp-server/rtsp-server-internal.h:
1473 * gst/rtsp-server/rtsp-stream-transport.c:
1474 * gst/rtsp-server/rtsp-stream.c:
1475 rtsp-stream: properly protect TCP backlog access
1477 We cannot hold stream->lock while pushing data, but need
1478 to consistently check the state of the backlog both from
1479 the send_tcp_message function and the on_message_sent function,
1480 which may or may not be called from the same thread.
1481 This commit introduces internal API to allow for potentially
1482 recursive locking of transport streams, addressing a race
1483 condition where the RTSP stream could push items out of order
1484 when popping them from the backlog.
1486 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1488 * gst/rtsp-server/rtsp-media.c:
1489 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1490 It's taken ownership of by the media, and returned with `transfer none`
1491 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1492 first then any bindings will wrongly take ownership of the pipeline once
1493 it arrives in bindings code.
1495 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1497 * examples/test-onvif-client.c:
1498 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1500 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1502 * gst/rtsp-server/rtsp-media.c:
1503 rtsp-media: fix default latency
1505 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1507 * gst/rtsp-server/rtsp-client.c:
1508 rtsp-client: make closing more thread safe
1509 + Take the watch lock prior to using priv->watch
1510 + Flush both the watch and connection before closing / unreffing
1511 gst_rtsp_connection_close() is not threadsafe on its own, this is
1512 a workaround at the client level, where we control both the watch
1515 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1517 * gst/rtsp-server/rtsp-latency-bin.c:
1518 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1521 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1522 `your_type_get_instance_private()` function instead
1525 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1527 * gst/rtsp-server/rtsp-client.c:
1528 * tests/check/gst/rtspserver.c:
1529 rtsp-client: add property post-session-timeout
1530 This is a TCP connection timeout for client connections, in seconds.
1531 If a positive value is set for this property, the client connection
1532 will be kept alive for this amount of seconds after the last session
1533 timeout. For negative values of this property the connection timeout
1534 handling is delegated to the system (just as it was before).
1537 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1539 * gst/rtsp-server/rtsp-stream.c:
1540 rtsp-stream: check for NULL transports prior to ref'ing
1542 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1544 * gst/rtsp-server/rtsp-server-internal.h:
1545 * gst/rtsp-server/rtsp-stream-transport.c:
1546 * gst/rtsp-server/rtsp-stream.c:
1547 rtsp-stream: fix checking of TCP backpressure
1548 The internal index of our appsinks, while it can be used to
1549 determine whether a message is RTP or RTCP, is not necessarily
1550 the same as the interleaved channel. Let the stream-transport
1551 determine the channel to check backpressure for, the same way
1552 it determines the channel according to whether it is sending
1555 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1557 * gst/rtsp-server/rtsp-session.c:
1558 rtsp-session: Butcher the file to please gst-indent in the CI
1559 This should be reverted once the CI has an updated gst-indent.
1561 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1563 * gst/rtsp-server/rtsp-session.c:
1564 * gst/rtsp-server/rtsp-session.h:
1565 * gst/rtsp-sink/gstrtspclientsink.c:
1566 * gst/rtsp-sink/gstrtspclientsink.h:
1567 rtsp-session & client: Remove deprecated GTimeVal
1568 GTimeVal won't work past 2038
1570 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1572 * gst/rtsp-server/rtsp-auth.c:
1573 rtsp-auth: fix default token leak
1575 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1577 * gst/rtsp-sink/gstrtspclientsink.c:
1578 gstrtspclientsink: unref transports when closing bin
1581 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1583 * gst/rtsp-server/rtsp-media.c:
1584 rtsp-media: Force seek when flush flag is set
1585 The commit "rtsp-client: define all seek accuracy flags from
1586 setup_play_mode" changed the behaviour of when doing a seek.
1587 Before that commit, having the flush flag set would result in a seek
1589 Even if no seek was needed. One reason to force seek is to flush old buffers
1590 created in Describe requests.
1591 Thus adding force seek also for flush flag will result in play request
1594 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1596 * gst/rtsp-server/rtsp-client.c:
1597 rtsp-client: Revitalize dead code
1598 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1601 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1603 * gst/rtsp-server/rtsp-sdp.c:
1604 rtsp-sdp: Don't try to use non-initialized values
1605 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1606 returns TRUE. Also avoid the whole clock signalling block if we're not
1607 dealing with senders.
1612 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1614 * gst/rtsp-server/rtsp-stream-transport.c:
1615 * gst/rtsp-server/rtsp-stream.c:
1616 * tests/check/gst/stream.c:
1617 rtsp-stream: Removing invalid transports returns false
1618 When removing transports an assertion was that the transports passed in
1619 for removal are present in the list, however that can't be assumed.
1620 As an example if a transport was removed from a thread running
1621 send_tcp_message, the main thread can try to remove the same transport
1622 again if it gets a handle_pause_request. This will not effect the
1623 transport list but it will effect n_tcp_transports as it will be
1624 decrement and then have the wrong value.
1626 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1628 * tests/check/gst/client.c:
1629 client test: add scale and speed negative tests
1630 Negative tests for scale and speed should be done as well, verify that
1631 the response code is "400 Bad request" when a bad request is done.
1633 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1635 * gst/rtsp-server/rtsp-auth.c:
1636 * gst/rtsp-server/rtsp-client.c:
1637 * gst/rtsp-server/rtsp-media-factory.c:
1638 * gst/rtsp-server/rtsp-media.c:
1639 * gst/rtsp-server/rtsp-server.c:
1640 * gst/rtsp-server/rtsp-session-pool.c:
1641 * gst/rtsp-server/rtsp-stream.c:
1642 * gst/rtsp-sink/gstrtspclientsink.c:
1643 Don't pass default GLib marshallers for signals
1644 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1645 actually internally optimize the signal (if the marshaller is available
1646 in GLib itself) by also setting the valist marshaller. This makes the
1647 signal emission a bit more performant than the regular marshalling,
1648 which still needs to box into `GValue` and call libffi in case of a
1650 Note that for custom marshallers, one would use
1651 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1653 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1655 * gst/rtsp-server/rtsp-mount-points.c:
1656 GstRTSPMountPoints: Remove any existing factory before adding a new one
1657 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1658 previous mount point will be freed" which was true when it was
1659 implemented using a GHashTable. But in 2012 it got rewrote using a
1660 GSequence and since then it could have 2 factories for the same path.
1661 Which one gets used is random, depending on the sorting order of 2
1664 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1666 * gst/rtsp-server/rtsp-client.c:
1667 * gst/rtsp-server/rtsp-server-internal.h:
1668 * gst/rtsp-server/rtsp-stream-transport.c:
1669 * gst/rtsp-server/rtsp-stream-transport.h:
1670 * gst/rtsp-server/rtsp-stream.c:
1671 stream: refactor TCP backpressure handling
1672 The previous implementation stopped sending TCP messages to
1673 all clients when a single one stopped consuming them, which
1674 obviously created problems for shared media.
1675 Instead, we now manage a backlog in stream-transport, and slow
1676 clients are removed once this backlog exceeds a maximum duration,
1677 currently hardcoded.
1680 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1683 meson: build gir even when cross-compiling if introspection was enabled explicitly
1684 This can be made to work in certain circumstances when
1685 cross-compiling, so default to not building g-i stuff
1686 when cross-compiling, but allow it if introspection was
1687 enabled explicitly via -Dintrospection=enabled.
1688 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1690 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1692 * gst/rtsp-server/rtsp-session.c:
1693 rtsp-session: clean up comment extra-timeout
1695 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1697 * gst/rtsp-server/rtsp-client.c:
1698 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1699 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1700 from the RTSP context.
1703 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1705 * gst/rtsp-server/rtsp-client.c:
1706 * gst/rtsp-server/rtsp-media.c:
1707 * gst/rtsp-server/rtsp-media.h:
1708 rtsp-client: Lock shared media
1709 For shared media we got race conditions. Concurrently rtsp clients might
1710 suspend or unsuspend the shared media and thus change the state without
1711 the clients expecting that.
1712 By introducing a lock that can be taken by callers such as rtsp_client
1713 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1714 to handle the media sequentially thus allowing one client to finish its
1715 rtsp call before another client calls on the same media.
1716 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1719 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1721 * gst/rtsp-server/rtsp-session.c:
1722 rtsp-session: add property extra-timeout
1723 Extra time to add to the timeout, in seconds. This only
1724 affects the time until a session is considered timed out
1725 and is not signalled in the RTSP request responses.
1726 Only the value of the timeout property is signalled in the
1729 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1731 * gst/rtsp-server/rtsp-stream.c:
1732 rtsp-stream : fix race condition in send_tcp_message
1733 If one thread is inside the send_tcp_message function and are done
1734 sending rtp or rtcp messages so the n_outstanding variable is zero
1735 however have not exit the loop sending the messages. While sending its
1736 messages, transports have been added or removed to the transport list,
1737 so the cache should be updated. If now an additional thread comes to
1738 the function send_tcp_message and trying to send rtp messages it will
1739 first destroy the rtp cache that is still being iterated trough by the
1743 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1752 * examples/.gitignore:
1753 * examples/Makefile.am:
1755 * gst/rtsp-server/.gitignore:
1756 * gst/rtsp-server/Makefile.am:
1757 * gst/rtsp-sink/Makefile.am:
1758 * pkgconfig/.gitignore:
1759 * pkgconfig/Makefile.am:
1761 * tests/Makefile.am:
1762 * tests/check/Makefile.am:
1763 Remove autotools build
1765 Maybe we can now use the meson pkgconfig module
1766 for .pc files? (Does it support uninstalled now?)
1768 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1770 * tests/check/gst/client.c:
1771 client: fix test mem leak in attach_rate_tweaking_probe
1773 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1775 * tests/check/gst/media.c:
1776 media: remove memleak in test test_media_seek
1778 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1780 * tests/check/gst/rtspserver.c:
1781 rtspserver: Remove memleak in test test_double_play
1783 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1785 * gst/rtsp-server/rtsp-media.c:
1786 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1788 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1790 * gst/rtsp-server/rtsp-media.c:
1791 * tests/check/gst/rtspserver.c:
1792 rtsp-media: Unblock all streams
1793 When unsuspending and going to PLAYING, unblock all streams instead of
1794 only those that are linked (the linked streams are the ones for which
1795 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1796 pushing buffers on unlinked streams.
1797 This change is because playback using single-threaded demuxers like
1798 matroska-demux could be blocked if SETUP was not called for all media.
1799 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1800 gstflvdemux, qtdemux, and matroska-demux) will handle
1801 GST_FLOW_NOT_LINKED automatically.
1804 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1806 * gst/rtsp-server/rtsp-media.c:
1807 * tests/check/gst/rtspserver.c:
1808 rtsp-media: Wait on async when needed.
1809 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1810 In the unit test the pause from adjust_play_mode will cause a preroll
1811 and after that async-done will be produced.
1812 Without this patch there are no one consuming this async-done and when
1813 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1814 wait for async-done. But then it wrongly find the async-done prodused by
1815 adjus_play_mode and continue executing without waiting for the preroll
1818 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1820 * gst/rtsp-server/rtsp-client.c:
1821 rtsp-client: RTP Info when completed_sender
1822 Change condition that should be fulfilled regarding RTPInfo.
1823 Replace !gst_rtsp_media_is_receive_only with
1824 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1825 for a sender pipeline that is complete. Only then a RTPInfo should
1827 gst_rtsp_media_is_receive_only gives different answears depending on
1829 If Describe is called wth URL+options for backchannel SDP will give only
1830 audio and only backchannel a=sendonly
1831 If Describe is called on URL+options that gives both audio and video
1832 direction from server to client, pipelines are created. Thus
1833 receive_only will return false, even though Setup only would setup
1835 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1836 streams are complete.
1838 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1840 * gst/rtsp-server/rtsp-client.c:
1841 * tests/check/gst/client.c:
1842 rtsp-client: RTP Info exists conditionally in PLAY
1843 If RTP Info is missing and it is not a receiver only, eg. audio
1844 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1845 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1846 Since 1.14 there is audio backchannel support. Thus RTP-info is
1847 conditional now. When audio backchannel only mode, there is no RTP-info.
1850 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1852 * examples/test-onvif-client.c:
1853 test-onvif-client: remove unused query
1855 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1857 * gst/rtsp-server/rtsp-client.c:
1858 rtsp-client: RTP Info must exist in PLAY response
1859 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1862 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1864 * examples/test-onvif-client.c:
1865 test-onvif-client: perform accurate seeks
1866 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1867 Also, modify how we compute the position: position queries in
1868 PAUSED mode fail to account for the newly-prerolled frame, leading
1869 to frame skips when performing seeks in that state. Instead,
1870 compute the current position from the last sample.
1872 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1874 * gst/rtsp-server/rtsp-client.c:
1875 * gst/rtsp-server/rtsp-media.c:
1876 * gst/rtsp-server/rtsp-media.h:
1877 * tests/check/gst/rtspserver.c:
1878 Use complete streams for scale and speed.
1879 Without this patch it's always stream0 that is used to get segment event
1880 that is used to set scale and speed. This even if client not doing SETUP
1881 for stream0. At least in suspend mode reset this not working since then
1882 it's just random if send_rtp_sink have got any segment event. There are
1883 no check if send_rtp_sink for stream0 got any data before media is
1884 prerolled after PLAY request.
1886 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1888 * examples/test-onvif-server.c:
1889 * examples/test-onvif-server.h:
1890 examples/onvif-server: fix werror build with clang
1891 ../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]
1892 self->incoming_segment->format, self->incoming_segment->flags,
1893 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1894 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1895 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
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>:77:1: note: expanded from here
1903 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1904 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1906 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1907 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1909 <scratch space>:9:1: note: expanded from here
1912 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1913 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1914 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1916 <scratch space>:12:1: note: expanded from here
1920 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1923 meson: Don't generate doc cache when no plugins are enabled
1924 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1926 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1928 * examples/test-onvif-client.c:
1929 test-onvif-client: stdin is not defined in MSVC
1931 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1933 * gst/rtsp-server/rtsp-media.c:
1934 rtsp-media: add missing Since tag
1936 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1938 * examples/test-onvif-client.c:
1939 test-onvif-client: STDIN_FILENO is not portable
1940 If not defined, define it to _fileno(stdin) on Windows, 0
1943 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1945 * examples/test-onvif-server.c:
1946 test-onvif-server: downgrade logging
1948 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1950 * examples/meson.build:
1951 * examples/test-onvif-client.c:
1952 * examples/test-onvif-server.c:
1953 examples: add ONVIF client / server example
1955 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1957 * gst/rtsp-server/rtsp-client.c:
1958 * gst/rtsp-server/rtsp-media.c:
1959 rtsp-client: define all seek accuracy flags from setup_play_mode
1960 We then pass those to adjust_play_mode, which needs to operate
1961 on the "final" seek flags, as previously the code in rtsp-media
1962 was assuming that accuracy seek flags (accurate / key_unit) should
1963 not be set if the flags passed to the seek method were already set.
1965 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1967 * gst/rtsp-server/rtsp-media-factory-uri.c:
1968 * gst/rtsp-server/rtsp-media.c:
1969 rtsp-media: Try to get dynamic payloaders by name from their bin first
1970 First try "pay", then "pay_%s" (where %s == pad name). And only then
1971 fall back to the code that simply takes the first payloader that is
1973 The current code usually works (but is racy) because it will always take
1974 the payloader that was last added (due to g_list_prepend() when adding
1975 elements) in pad-added and that's usually the correct one. But if a new
1976 payloader is added between pad-added and us trying to get it, we would
1977 get the wrong payloader.
1979 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1981 * tests/check/gst/client.c:
1982 client test: expect any port in transport
1983 setup_multicast_client sets a 5000-5010 range for the client
1984 ports, it is incorrect to expect the transport to always use
1988 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1990 * tests/check/gst/onvif.c:
1991 onvif tests: use g_cond_wait() correctly
1992 g_cond_wait() has to be called in a loop until required conditions
1996 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1998 * gst/rtsp-server/rtsp-stream.c:
1999 rtsp-stream: Not wait on receiver streams when pre-rolling
2000 Without this patch there are problem pre-rolling when using audio back
2002 Without this patch a probe will be created for all streams including
2003 the stream for audio backchannel. To pre-roll all this pads have to
2004 receive data. Since the stream for audio backchannel is a receiver this
2006 The solution is to never create any probes for streams that are for
2007 incomming data and instead set them as blocking already from beginning.
2009 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
2011 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2012 * gst/rtsp-server/rtsp-onvif-media.c:
2013 onvif-media: fix "void function returning a value" compiler warning
2015 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2017 * gst/rtsp-server/rtsp-media.c:
2018 rtsp-media: make sure streams are blocked when sending seek
2019 The recent ONVIF work exposed a race condition when dealing with
2020 multiple streams: one of the sinks may preroll before other streams
2021 have started flushing. This led to the pipeline posting async-done
2022 prematurely, when some streams were actually still in the middle
2023 of performing a flushing seek. The newly-added code looks up a
2024 sticky segment event on the first stream in order to respond to
2025 the PLAY request with accurate Scale and Speed headers. In the
2026 failure condition, the first stream was flushing, and thus had
2027 no sticky segment event, leading to the PLAY request failing,
2028 and in turn the test.
2030 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
2033 * gst/rtsp-server/rtsp-media-factory-uri.h:
2036 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2038 * gst/rtsp-server/rtsp-client.c:
2039 * gst/rtsp-server/rtsp-client.h:
2040 * gst/rtsp-server/rtsp-media.c:
2041 * gst/rtsp-server/rtsp-media.h:
2042 * gst/rtsp-server/rtsp-onvif-client.c:
2043 * gst/rtsp-server/rtsp-onvif-client.h:
2044 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2045 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2046 * gst/rtsp-server/rtsp-onvif-media.c:
2047 * gst/rtsp-server/rtsp-onvif-server.h:
2048 * gst/rtsp-server/rtsp-stream.c:
2049 * gst/rtsp-server/rtsp-stream.h:
2050 * tests/check/gst/media.c:
2051 * tests/check/gst/onvif.c:
2052 * tests/check/meson.build:
2053 onvif: Implement and test the Streaming Specification
2054 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2056 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2058 * gst/rtsp-server/rtsp-client.c:
2059 * gst/rtsp-server/rtsp-client.h:
2060 rtsp-client: add gst_rtsp_client_get_stream_transport()
2061 This will be used in the onvif tests in order to validate the
2062 data transmitted over TCP: for streaming to continue after a
2063 data message has been provided to client->send_func, the client
2064 is responsible for marking the message as sent on the relevant
2067 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2069 * gst/rtsp-server/rtsp-client.c:
2070 client: Scale implies TRICK_MODE
2072 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2074 * gst/rtsp-server/rtsp-client.c:
2075 client: compare booleans, not pointers to them
2077 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
2079 * gst/rtsp-server/rtsp-media.c:
2080 * gst/rtsp-server/rtsp-stream.c:
2081 * tests/check/gst/media.c:
2082 Reverse playback support
2083 GStreamer plays segment from stop to start when doing reverse playback.
2084 RTSP implies that media should be played from start of Range header to
2085 its stop. Hence we swap start and stop times before passing them to
2087 Also make gst_rtsp_stream_query_stop always return value that can be
2088 used as stop time of Range header.
2090 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
2092 * gst/rtsp-server/rtsp-client.c:
2093 * gst/rtsp-server/rtsp-media.c:
2094 * gst/rtsp-server/rtsp-media.h:
2095 * tests/check/gst/client.c:
2096 rtsp-client: add support for Scale and Speed header
2097 Add support for the RTSP Scale and Speed headers by setting the rate in
2098 the seek to (scale*speed). We then check the resulting segment for rate
2099 and applied rate, and use them as values for the Speed and Scale headers
2101 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2103 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
2105 * gst/rtsp-server/rtsp-client.c:
2106 * gst/rtsp-server/rtsp-client.h:
2107 rtsp-client: allow sub classes to adjust the seek
2108 Adds a new virtual function, adjust_play_mode(), that allows
2109 sub classes to adjust the seek done on the media. The sub class can
2110 modify the values of the the seek flags and the rate.
2111 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2113 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
2115 * gst/rtsp-server/rtsp-media.c:
2116 * gst/rtsp-server/rtsp-media.h:
2117 * gst/rtsp-server/rtsp-stream.c:
2118 * gst/rtsp-server/rtsp-stream.h:
2119 * tests/check/gst/media.c:
2120 rtsp-media: allow specifying rate when seeking
2121 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
2122 caller to specify the rate for the seek. Also added functions in
2123 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
2124 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2126 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
2130 meson: Bump minimal GLib version to 2.44
2131 This means we can use some newer features and get rid of some
2132 boilerplate code using the G_DECLARE_* macros.
2133 As discussed on IRC, 2.44 is old enough by now to start depending on it.
2135 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2137 * docs/libs/.gitignore:
2138 * docs/libs/Makefile.am:
2139 * docs/libs/gst-rtsp-server-docs.sgml:
2140 * docs/libs/gst-rtsp-server-sections.txt:
2141 * docs/libs/gst-rtsp-server.types:
2142 docs: remove obsolete gtk-doc related files
2144 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2146 * gst/rtsp-sink/gstrtspclientsink.c:
2147 doc: remove xml from comments
2149 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
2151 * docs/gst_plugins_cache.json:
2153 docs: Stop building the doc cache by default
2154 And update the cache
2155 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
2157 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
2159 * docs/gst_plugins_cache.json:
2160 docs: Update plugins documentation cache
2162 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
2165 * gst/rtsp-server/rtsp-context.c:
2166 * gst/rtsp-server/rtsp-session-pool.c:
2167 doc: Fix some docstrings
2169 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
2175 * docs/gst_plugins_cache.json:
2178 * docs/plugin-index.md:
2179 * docs/plugin-sitemap.txt:
2182 * docs/version.entities.in:
2183 * gst/rtsp-server/meson.build:
2184 * gst/rtsp-sink/meson.build:
2186 * meson_options.txt:
2187 docs: Port to hotdoc
2189 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2191 * gst/rtsp-server/rtsp-auth.c:
2192 * gst/rtsp-server/rtsp-client.h:
2193 rtsp-server: Fix various Since markers
2195 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
2197 * gst/rtsp-server/rtsp-media.c:
2198 * gst/rtsp-server/rtsp-sdp.c:
2199 * gst/rtsp-server/rtsp-session-media.c:
2200 * gst/rtsp-server/rtsp-stream.c:
2201 rtsp-server: Add various Since: 1.14 markers
2203 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
2205 * gst/rtsp-server/rtsp-media-factory.c:
2206 * gst/rtsp-server/rtsp-media.c:
2207 * gst/rtsp-server/rtsp-stream-transport.c:
2208 * gst/rtsp-server/rtsp-stream.c:
2209 rtsp-server: Add various missing Since: 1.16 markers
2211 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
2213 * gst/rtsp-sink/gstrtspclientsink.c:
2214 rtspclientsink: Set async-handling=false for the internal bins
2215 Without this we can easily run into a race condition with async state changes:
2216 - the pipeline is doing an async state change
2217 - we set the internal bins to PLAYING but that's ignored because an
2218 async state change is currently pending
2219 - the async state change finishes but does not change the state of the
2220 internal bins because of locked_state==TRUE
2221 - the internal bins stay in PAUSED forever
2223 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2225 * gst/rtsp-sink/gstrtspclientsink.c:
2226 rtspclientsink: Use write_messages() API to send buffer lists in one go
2227 And to write messages with multiple memories also via writev().
2229 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
2231 * gst/rtsp-server/rtsp-client.c:
2232 * gst/rtsp-server/rtsp-client.h:
2233 * gst/rtsp-server/rtsp-server-object.h:
2234 * gst/rtsp-server/rtsp-server.c:
2235 rtsp-client: Handle Content-Length limitation
2236 Add functionality to limit the Content-Length.
2237 API addition, Enhancement.
2238 Define an appropriate request size limit and reject requests
2239 exceeding the limit with response status 413 Request Entity Too Large
2242 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2249 === release 1.16.0 ===
2251 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2257 * gst-rtsp-server.doap:
2261 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
2263 * gst/rtsp-sink/gstrtspclientsink.c:
2264 rtspclientsink: Notify the stream transport about each written message
2265 Otherwise it will never try to send us the next one: it tries to keep
2266 exactly one message in-flight all the time.
2267 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
2268 in the client sink we always write data out synchronously.
2270 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
2272 * gst/rtsp-server/rtsp-stream.c:
2273 rtsp_server: Free thread pool before clean transport cache
2274 If not waiting for free thread pool before clean transport caches, there
2275 can be a crash if a thread is executing in transport list loop in
2276 function send_tcp_message.
2277 Also add a check if priv->send_pool in on_message_sent to avoid that a
2278 new thread is pushed during wait of free thread pool. This is possible
2279 since when waiting for free thread pool mutex have to be unlocked.
2281 === release 1.15.90 ===
2283 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
2289 * gst-rtsp-server.doap:
2293 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
2295 * gst/rtsp-server/rtsp-stream.c:
2296 rtsp-stream: Add support for GCM (RFC 7714)
2299 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2301 * gst/rtsp-server/rtsp-session-pool.c:
2302 session pool: fix missing klass-> in klass->create_session
2304 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2307 g-i: pass --quiet to g-ir-scanner
2308 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2309 that we get even if everything works just fine.
2310 We still get g-ir-scanner warnings and compiler warnings if
2311 we pass this option.
2313 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2316 g-i: silence 'nested extern' compiler warnings when building scanner binary
2317 We need a nested extern in our init section for the scanner binary
2318 so we can call gst_init to make sure GStreamer types are initialised
2319 (they are not all lazy init via get_type functions, but some are in
2320 exported variables). There doesn't seem to be any other mechanism to
2321 achieve this, so just remove that warning, it's not important at all.
2323 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2326 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2328 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2330 * gst/rtsp-server/rtsp-media.c:
2331 * tests/check/gst/media.c:
2332 rtsp-media: Handle set state when preparing.
2333 Handle the situation when a call to gst_rtsp_media_set_state is done
2334 when media status is preparing.
2335 Also add unit test for this scenario.
2336 The unit test simulate on a media level when two clients share a (live)
2338 Both clients have done SETUP and got responses. Now client 1 is doing
2339 play and client 2 is just closing the connection.
2340 Then without patch there are a problem when
2341 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2342 And client2 is doing closing connection we can end up in a call
2343 to gst_rtsp_media_set_state when
2344 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2345 shut down media is jumped over .
2346 With this patch and this scenario we wait until
2347 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2348 execute after that and now we will execute the logic for
2351 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2359 === release 1.15.2 ===
2361 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2367 * gst-rtsp-server.doap:
2371 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2373 * gst/rtsp-server/rtsp-media.c:
2374 * tests/check/gst/client.c:
2375 rtsp-media: Fix multicast use case with common media
2384 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2386 * gst/rtsp-server/rtsp-client.c:
2387 * gst/rtsp-server/rtsp-stream.c:
2388 * gst/rtsp-server/rtsp-stream.h:
2389 rtsp-server: remove recursive behavior
2390 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2392 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2394 * gst/rtsp-server/rtsp-client.c:
2395 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2396 And route all messages through the send_func if no send_messages_func
2398 We otherwise break backwards compatibility.
2400 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2402 * docs/libs/gst-rtsp-server-sections.txt:
2403 * gst/rtsp-server/rtsp-client.c:
2404 * gst/rtsp-server/rtsp-client.h:
2405 * gst/rtsp-server/rtsp-stream.c:
2406 rtsp-client: Add support for sending buffer lists directly
2407 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2409 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2411 * docs/libs/gst-rtsp-server-sections.txt:
2412 * gst/rtsp-server/rtsp-client.c:
2413 * gst/rtsp-server/rtsp-media.c:
2414 * gst/rtsp-server/rtsp-stream-transport.c:
2415 * gst/rtsp-server/rtsp-stream-transport.h:
2416 * gst/rtsp-server/rtsp-stream.c:
2417 * gst/rtsp-sink/gstrtspclientsink.c:
2418 rtsp-server: Add support for buffer lists
2419 This adds new functions for passing buffer lists through the different
2420 layers without breaking API/ABI, and enables the appsink to actually
2421 provide buffer lists.
2422 This should already reduce CPU usage and potentially context switches a
2423 bit by passing a whole buffer list from the appsink instead of
2424 individual buffers. As a next step it would be necessary to
2425 a) Add support for a vector of data for the GstRTSPMessage body
2426 b) Add support for sending multiple messages at once to the
2427 GstRTSPWatch and let it be handled internally
2428 c) Adding API to GOutputStream that works like writev()
2429 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2431 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2433 * gst/rtsp-server/rtsp-client.c:
2434 client: Fix crash in close handler
2435 The close handler could trigger a crash because it invalidated the
2436 watch_context while still leaving a source attached to it which would be
2437 cleaned up at a later point.
2439 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2441 * gst/rtsp-server/rtsp-stream.c:
2442 rtsp-stream: Use cached address when allocating sockets
2443 If an address/port was previously decided upon (ex: multicast in the
2444 SDP), then use that instead of re-creating another one
2445 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2447 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2449 * gst/rtsp-server/rtsp-media.c:
2450 rtsp-media: Fix race codition in finish_unprepare
2451 The previous fix for race condition around finish_unprepare where the
2452 function could be called twice assumed that the status wouldn't change
2453 during execution of the function. This assumption is incorrect as the
2454 state may change, for example if an error message arrives from the
2456 Instead a flag keeping track on whether the finish_unprepare function
2457 is currently executing is introduced and checked.
2458 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2460 === release 1.15.1 ===
2462 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2468 * gst-rtsp-server.doap:
2472 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2474 * gst/rtsp-server/rtsp-stream.c:
2475 Add source elements to the pipeline before activation
2476 In plug_src we changed the element state before adding it to
2477 the owner container. This prevented the pipeline from intercepting
2478 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2479 to assign a custom task pool.
2480 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2482 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2485 Automatic update of common submodule
2486 From ed78bee to 59cb678
2488 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2490 * examples/test-appsrc.c:
2491 examples: test-appsrc: fix coding style error
2493 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2495 * examples/test-appsrc.c:
2496 examples: test-appsrc: fix buffer leak
2498 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2500 * gst/rtsp-server/rtsp-media.c:
2501 rtsp-media: Update priv->blocked when linked streams are unblocked.
2502 Media is considered to be blocked when all streams that belong to
2503 that media are blocked.
2504 This patch solves the problem of inconsistent updates of
2505 priv->blocked that are not synchronized with the media state.
2507 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2509 * gst/rtsp-server/rtsp-media.c:
2510 rtsp-media: Don't block streams before seeking
2511 Before the seek operation is performed on media, it's required that
2512 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2513 At this stage, all transport parts (transport sinks) have been successfully
2514 added to the pipeline and there is no need for blocking the streams.
2516 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2518 * tests/check/gst/rtspserver.c:
2519 tests: rtspserver: Add shared media test case for TCP
2521 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2523 * gst/rtsp-server/rtsp-stream.c:
2524 rtsp-stream: Use seqnum-offset for rtpinfo
2525 The sequence number in the rtpinfo is supposed to be the first RTP
2526 sequence number. The "seqnum" property on a payloader is supposed to be
2527 the number from the last processed RTP packet. The sequence number for
2528 payloaders that inherit gstrtpbasepayload will not be correct in case of
2529 buffer lists. In order to fix the seqnum property on the payloaders
2530 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2531 "seqnum-offset" from the "stats" property contains the value of the
2532 very first RTP packet in a stream. The server will, however, try to look
2533 at the last simple in the sink element and only use properties on the
2534 payloader in case there no sink elements yet, and by looking at the last
2535 sample of the sink gives the server full control of which RTP packet it
2536 looks at. If the payloader does not have the "stats" property, "seqnum"
2537 is still used since "seqnum-offset" is only present in as part of
2538 "stats" and this is still an issue not solved with this patch.
2539 Needed for gst-plugins-base!17
2541 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2543 * gst/rtsp-server/rtsp-stream.c:
2544 rtsp-stream: Plug memory leak
2545 Attaching a GSource to a context will increase the refcount. The idle
2546 source will never be free'd since the initial reference is never
2549 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2552 Add Gitlab CI configuration
2553 This commit adds a .gitlab-ci.yml file, which uses a feature
2554 to fetch the config from a centralized repository. The intent is
2555 to have all the gstreamer modules use the same configuration.
2556 The configuration is currently hosted at the gst-ci repository
2557 under the gitlab/ci_template.yml path.
2558 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2560 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2563 * gst-rtsp-server.doap:
2564 Update git locations to gitlab
2566 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2568 * gst/rtsp-server/meson.build:
2569 meson: add new onvif types
2571 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2573 * gst/rtsp-server/meson.build:
2574 Add ONVIF subclass headers to the installed headers in meson.build too
2576 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2578 * gst/rtsp-server/rtsp-server-object.h:
2579 * gst/rtsp-server/rtsp-server.h:
2580 rtsp-server: Declare GstRTSPServer struct before anything else
2581 It's needed by all kinds of other headers, including the ones that are
2582 required for defining the GstRTSPServer struct itself and its API.
2584 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2586 * gst/rtsp-server/rtsp-onvif-client.h:
2587 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2588 * gst/rtsp-server/rtsp-onvif-media.h:
2589 * gst/rtsp-server/rtsp-onvif-server.h:
2590 Mark all ONVIF-specific subclasses as Since 1.14
2592 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2594 * gst/rtsp-server/Makefile.am:
2595 * gst/rtsp-server/meson.build:
2596 * gst/rtsp-server/rtsp-context.h:
2597 * gst/rtsp-server/rtsp-onvif-server.c:
2598 * gst/rtsp-server/rtsp-onvif-server.h:
2599 * gst/rtsp-server/rtsp-server-object.h:
2600 * gst/rtsp-server/rtsp-server-prelude.h:
2601 * gst/rtsp-server/rtsp-server.c:
2602 * gst/rtsp-server/rtsp-server.h:
2603 * gst/rtsp-server/rtsp-session.h:
2604 Include ONVIF types from single-include rtsp-server.h
2605 ... by actually making it a single-include header and moving everything
2606 related to the GstRTSPServer type to rtsp-server-object.h instead.
2607 Otherwise there are too many circular includes.
2608 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2610 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2612 * gst/rtsp-server/rtsp-client.c:
2613 * gst/rtsp-server/rtsp-latency-bin.c:
2614 * gst/rtsp-server/rtsp-stream.c:
2615 * gst/rtsp-server/rtsp-stream.h:
2616 rtsp-stream: use idle source in on_message_sent
2617 When the underlying layers are running on_message_sent, this sometimes
2618 causes the underlying layer to send more data, which will cause the
2619 underlying layer to run callback on_message_sent again. This can go on
2621 To break this chain, we introduce an idle source that takes care of
2622 sending data if there are more to send when running callback
2623 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2625 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2627 * gst/rtsp-server/rtsp-client.c:
2628 rtsp-client: Remove timeout GSource on cleanup
2629 Avoids ending up with races where a timeout would still be around
2630 *after* a client was gone. This could happen rather easily in
2631 RTSP-over-HTTP mode on a local connection, where each RTSP message
2632 would be sent as a different HTTP connection with the same tunnelid.
2633 If not properly removed, that timeout would then try to free again
2634 a client (and its contents).
2636 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2638 * gst/rtsp-server/Makefile.am:
2639 autotools: fix distcheck
2641 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2643 * gst/rtsp-server/Makefile.am:
2644 * gst/rtsp-server/meson.build:
2645 * gst/rtsp-server/rtsp-latency-bin.c:
2646 * gst/rtsp-server/rtsp-latency-bin.h:
2647 * gst/rtsp-server/rtsp-onvif-media.c:
2648 onvif: encapsulate onvif part into a bin
2649 ...and thus do not let onvif affect pipelines latency
2650 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2652 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2654 * tests/check/gst/client.c:
2655 tests: client: Avoid bind() failures in tests
2656 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2658 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2660 * gst/rtsp-server/rtsp-media-factory.c:
2661 * gst/rtsp-server/rtsp-media-factory.h:
2662 * gst/rtsp-server/rtsp-media.c:
2663 * gst/rtsp-server/rtsp-media.h:
2664 * gst/rtsp-server/rtsp-stream.c:
2665 * gst/rtsp-server/rtsp-stream.h:
2666 * tests/check/gst/client.c:
2667 * tests/check/gst/mediafactory.c:
2668 New property for socket binding to mcast addresses
2669 By default the multicast sockets are bound to INADDR_ANY,
2670 as it's not allowed to bind sockets to multicast addresses
2671 in Windows. This default behaviour can be changed by setting
2672 bind-mcast-address property on the media-factory object.
2673 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2675 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2678 * gst/rtsp-server/Makefile.am:
2679 * gst/rtsp-server/meson.build:
2680 * gst/rtsp-server/rtsp-address-pool.c:
2681 * gst/rtsp-server/rtsp-auth.c:
2682 * gst/rtsp-server/rtsp-client.c:
2683 * gst/rtsp-server/rtsp-context.c:
2684 * gst/rtsp-server/rtsp-media-factory-uri.c:
2685 * gst/rtsp-server/rtsp-media-factory.c:
2686 * gst/rtsp-server/rtsp-media.c:
2687 * gst/rtsp-server/rtsp-mount-points.c:
2688 * gst/rtsp-server/rtsp-params.c:
2689 * gst/rtsp-server/rtsp-permissions.c:
2690 * gst/rtsp-server/rtsp-sdp.c:
2691 * gst/rtsp-server/rtsp-server-prelude.h:
2692 * gst/rtsp-server/rtsp-server.c:
2693 * gst/rtsp-server/rtsp-session-media.c:
2694 * gst/rtsp-server/rtsp-session-pool.c:
2695 * gst/rtsp-server/rtsp-session.c:
2696 * gst/rtsp-server/rtsp-stream-transport.c:
2697 * gst/rtsp-server/rtsp-stream.c:
2698 * gst/rtsp-server/rtsp-thread-pool.c:
2699 * gst/rtsp-server/rtsp-token.c:
2701 libs: fix API export/import and 'inconsistent linkage' on MSVC
2702 Export rtsp-server library API in headers when we're building the
2703 library itself, otherwise import the API from the headers.
2704 This fixes linker warnings on Windows when building with MSVC.
2705 Fix up some missing config.h includes when building the lib which
2706 is needed to get the export api define from config.h
2707 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2709 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2711 * gst/rtsp-server/rtsp-media-factory.c:
2712 rtsp-media-factory: Add missing break statements
2713 This resulted in warnings/assertions whenever one accessed the
2714 max-mcast-ttl property.
2718 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2721 * meson_options.txt:
2722 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2724 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2727 * meson_options.txt:
2728 * tests/check/meson.build:
2729 meson: add option to disable build of rtspclientsink plugin
2731 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2733 * meson_options.txt:
2734 meson: re-arrange options
2736 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2739 * meson_options.txt:
2740 * tests/check/meson.build:
2741 * tests/meson.build:
2742 meson: Use feature option for tests option
2743 This was somehow missed the last time around.
2745 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2747 * gst/rtsp-server/meson.build:
2749 meson: Maintain macOS ABI through dylib versioning
2750 Requires Meson 0.48, but the feature will be ignored on older versions
2751 so it's safe to add it without bumping the requirement.
2753 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2755 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2757 * gst/rtsp-sink/meson.build:
2759 meson: add pkg-config file for the rtspclientsink plugin
2761 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2763 * gst/rtsp-server/rtsp-client.c:
2764 * tests/check/gst/client.c:
2765 rtsp-client: Avoid reuse of channel numbers for interleaved
2766 If a (strange) client would reuse interleaved channel numbers in
2767 multiple SETUP requests, we should not accept them. The channel
2768 numbers are used for looking up stream transports in the
2769 priv->transports hash table, and transports disappear from the table
2770 if channel numbers are reused.
2771 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2772 server to change the channel numbers suggested by the client.
2773 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2775 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2777 * tests/check/gst/client.c:
2778 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2779 Allow regex for matching transport header against expected pattern.
2780 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2782 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2784 * tests/check/meson.build:
2785 meson: There is no gstreamer-plugins-good-1.0.pc
2786 There is no installed version of that, only an uninstalled version.
2788 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2790 * gst/rtsp-server/rtsp-client.c:
2791 * tests/check/gst/stream.c:
2792 Fix indentation again
2794 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2796 * gst/rtsp-server/rtsp-client.c:
2797 * gst/rtsp-server/rtsp-stream.c:
2798 * gst/rtsp-server/rtsp-stream.h:
2799 * tests/check/gst/client.c:
2800 * tests/check/gst/stream.c:
2801 stream: Added a list of multicast client addresses
2802 When media is shared, the same media stream can be sent
2803 to multiple multicast groups. Currently, there is no API
2804 to retrieve multicast addresses from the stream.
2805 When calling gst_rtsp_stream_get_multicast_address() function,
2806 only the first multicast address is returned.
2807 With this patch, each multicast destination requested in SETUP
2808 will be stored in an internal list (call to
2809 gst_rtsp_stream_add_multicast_client_address()).
2810 The list of multicast groups requested by the clients can be
2811 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2812 There still exist some problems with the current implementation
2813 in the multicast case:
2814 1) The receiving part is currently only configured with
2815 regard to the first multicast client (see
2816 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2817 2) Secondly, of security reasons, some constraints should be
2818 put on the requested multicast destinations (see
2819 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2820 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2821 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2823 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2825 * gst/rtsp-server/rtsp-client.c:
2826 * gst/rtsp-server/rtsp-stream.c:
2827 * gst/rtsp-server/rtsp-stream.h:
2828 * tests/check/gst/client.c:
2829 stream: Choose the maximum ttl value provided by multicast clients
2830 The maximum ttl value provided so far by the multicast clients
2831 will be chosen and reported in the response to the current
2833 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2834 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2836 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2838 * gst/rtsp-server/rtsp-stream.c:
2839 * tests/check/gst/client.c:
2840 rtsp-stream: Don't require address pool in the transport specific case
2841 If "transport.client-settings" parameter is set to true, the client is
2842 allowed to specify destination, ports and ttl.
2843 There is no need for pre-configured address pool.
2844 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2845 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2847 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2849 * gst/rtsp-server/rtsp-client.c:
2850 * tests/check/gst/client.c:
2851 client: Don't reserve multicast address in the client setting case
2852 When two multicast clients request specific transport
2853 configurations, and "transport.client-settings" parameter is
2854 set to true, it's wrong to actually require that these two
2855 clients request the same multicast group.
2856 Removed test_client_multicast_invalid_transport_specific test
2857 cases as they wrongly require that the requested destination
2858 address is supposed to be present in the address pool, also in
2859 the case when "transport.client-settings" parameter is set to true.
2860 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2861 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2863 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2865 * gst/rtsp-server/rtsp-media-factory.c:
2866 * gst/rtsp-server/rtsp-media-factory.h:
2867 * gst/rtsp-server/rtsp-media.c:
2868 * gst/rtsp-server/rtsp-media.h:
2869 * gst/rtsp-server/rtsp-stream.c:
2870 * gst/rtsp-server/rtsp-stream.h:
2871 * tests/check/gst/mediafactory.c:
2872 Add new API for setting/getting maximum multicast ttl value
2873 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2874 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2876 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2878 * gst/rtsp-server/rtsp-stream.c:
2879 rtsp-stream: avoid duplicating the first multicast client
2880 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2881 clients were dynamically added and removed to the multicast
2882 udp sinks, as such we should no longer add a first client in
2883 set_multicast_socket_for_udpsink
2884 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2886 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2888 * gst/rtsp-server/rtsp-stream.c:
2889 Revert "rtsp-stream: avoid duplicating the first multicast client"
2890 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2891 Commits where accidentially squashed together
2893 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2895 * gst/rtsp-server/rtsp-client.c:
2896 * gst/rtsp-server/rtsp-media-factory.c:
2897 * gst/rtsp-server/rtsp-media-factory.h:
2898 * gst/rtsp-server/rtsp-media.c:
2899 * gst/rtsp-server/rtsp-media.h:
2900 * gst/rtsp-server/rtsp-stream.c:
2901 * gst/rtsp-server/rtsp-stream.h:
2902 * tests/check/gst/client.c:
2903 * tests/check/gst/mediafactory.c:
2904 Revert "Add new API for setting/getting maximum multicast ttl value"
2905 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2906 Commits where accidentially squashed together
2908 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2910 * gst/rtsp-server/rtsp-stream.c:
2911 * tests/check/gst/client.c:
2912 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2913 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2914 Commits where accidentially squashed together
2916 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.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 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2924 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2925 Commits where accidentially squashed together
2927 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2929 * examples/test-auth-digest.c:
2930 examples: Fix indentation
2932 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2934 * gst/rtsp-server/rtsp-client.c:
2935 * gst/rtsp-server/rtsp-stream.c:
2936 * gst/rtsp-server/rtsp-stream.h:
2937 * tests/check/gst/client.c:
2938 * tests/check/gst/stream.c:
2939 stream: Choose the maximum ttl value provided by multicast clients
2940 The maximum ttl value provided so far by the multicast clients
2941 will be chosen and reported in the response to the current
2943 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2945 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2947 * gst/rtsp-server/rtsp-stream.c:
2948 * tests/check/gst/client.c:
2949 rtsp-stream: Don't require address pool in the transport specific case
2950 If "transport.client-settings" parameter is set to true, the client is
2951 allowed to specify destination, ports and ttl.
2952 There is no need for pre-configured address pool.
2953 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2955 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2957 * gst/rtsp-server/rtsp-client.c:
2958 * gst/rtsp-server/rtsp-media-factory.c:
2959 * gst/rtsp-server/rtsp-media-factory.h:
2960 * gst/rtsp-server/rtsp-media.c:
2961 * gst/rtsp-server/rtsp-media.h:
2962 * gst/rtsp-server/rtsp-stream.c:
2963 * gst/rtsp-server/rtsp-stream.h:
2964 * tests/check/gst/client.c:
2965 * tests/check/gst/mediafactory.c:
2966 Add new API for setting/getting maximum multicast ttl value
2967 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2969 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2971 * gst/rtsp-server/rtsp-stream.c:
2972 rtsp-stream: avoid duplicating the first multicast client
2973 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2974 clients were dynamically added and removed to the multicast
2975 udp sinks, as such we should no longer add a first client in
2976 set_multicast_socket_for_udpsink
2977 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2979 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2981 * gst/rtsp-server/Makefile.am:
2982 rtsp-server: Add gstreamer-base gir dir in autotools
2984 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2986 * gst/rtsp-server/rtsp-client.c:
2987 * gst/rtsp-server/rtsp-stream.c:
2988 rtsp-client: always allocate both IPV4 and IPV6 sockets
2989 multiudpsink does not support setting the socket* properties
2990 after it has started, which meant that rtsp-server could no
2991 longer serve on both IPV4 and IPV6 sockets since the patches
2992 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2994 When first connecting an IPV6 client then an IPV4 client,
2995 multiudpsink fell back to using the IPV6 socket.
2996 When first connecting an IPV4 client, then an IPV6 client,
2997 multiudpsink errored out, released the IPV4 socket, then
2998 crashed when trying to send a message on NULL nevertheless,
2999 that is however a separate issue.
3000 This could probably be fixed by handling the setting of
3001 sockets in multiudpsink after it has started, that will
3002 however be a much more significant effort.
3003 For now, this commit simply partially reverts the behaviour
3004 of rtsp-stream: it will continue to only create the udpsinks
3005 when needed, as was the case since the patches were merged,
3006 it will however when creating them, always allocate both
3007 sockets and set them on the sink before it starts, as was
3008 the case prior to the patches.
3009 Transport configuration will only error out if the allocation
3010 of UDP sockets fails for the actual client's family, this
3011 also downgrades the GST_ERRORs in alloc_ports_one_family
3012 to GST_WARNINGs, as failing to allocate is no longer
3014 https://bugzilla.gnome.org/show_bug.cgi?id=796875
3016 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3019 * meson_options.txt:
3020 meson: Convert common options to feature options
3021 These are necessary for gst-build to set options correctly. The
3022 remaining automagic option is cgroup support in examples.
3023 https://bugzilla.gnome.org/show_bug.cgi?id=795107
3025 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
3027 * gst/rtsp-server/rtsp-stream.c:
3028 rtsp-stream: Slightly simplify locking
3030 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
3032 * gst/rtsp-server/rtsp-client.c:
3033 * gst/rtsp-server/rtsp-stream-transport.c:
3034 * gst/rtsp-server/rtsp-stream-transport.h:
3035 * gst/rtsp-server/rtsp-stream.c:
3036 Limit queued TCP data messages to one per stream
3037 Before, the watch backlog size in GstRTSPClient was changed
3038 dynamically between unlimited and a fixed size, trying to avoid both
3039 unlimited memory usage and deadlocks while waiting for place in the
3040 queue. (Some of the deadlocks were described in a long comment in
3042 In the previous commit, we changed to a fixed backlog size of 100.
3043 This is possible, because we now handle RTP/RTCP data messages differently
3044 from RTSP request/response messages.
3045 The data messages are messages tunneled over TCP. We allow at most one
3046 queued data message per stream in GstRTSPClient at a time, and
3047 successfully sent data messages are acked by sending a "message-sent"
3048 callback from the GstStreamTransport. Until that ack comes, the
3049 GstRTSPStream does not call pull_sample() on its appsink, and
3050 therefore the streaming thread in the pipeline will not be blocked
3051 inside GstRTSPClient, waiting for a place in the queue.
3052 pull_sample() is called when we have both an ack and a "new-sample"
3053 signal from the appsink. Then, we know there is a buffer to write.
3054 RTSP request/response messages are not acked in the same way as data
3055 messages. The rest of the 100 places in the queue are used for
3056 them. If the queue becomes full of request/response messages, we
3057 return an error and close the connection to the client.
3058 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
3060 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
3062 * gst/rtsp-server/rtsp-client.c:
3063 rtsp-client: Use fixed backlog size
3064 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
3065 Preparation for the next commit, which changes to a different way of
3066 avoiding both deadlocks and unlimited memory usage with the watch
3069 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3071 * gst/rtsp-server/rtsp-media.c:
3072 rtsp-media: unref clock (if set) when finalizing
3073 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3075 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3077 * docs/libs/gst-rtsp-server-sections.txt:
3078 rtsp-media: add gst_rtsp_media_*_set_clock to docs
3079 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3081 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
3083 * gst/rtsp-server/rtsp-media-factory.c:
3084 media-factory: unref old clock when setting new clock
3085 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3087 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
3089 * gst/rtsp-server/rtsp-media-factory.c:
3090 media-factory: unref clock in finalize
3091 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3093 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
3095 * gst/rtsp-server/rtsp-onvif-media.c:
3096 rtsp-onvif-media: fix g-ir-scanner warnings
3098 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3101 .gitignore: add another example binary
3103 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
3105 * examples/meson.build:
3106 meson: add new test-appsrc2 example to meson build
3108 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
3110 * examples/Makefile.am:
3111 examples: fix build of new test-appsrc2 example
3112 Need to link against libgstapp-1.0.
3114 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
3116 * examples/.gitignore:
3117 * examples/Makefile.am:
3118 * examples/test-appsrc2.c:
3119 examples: Add test-appsrc2
3120 Add an example of feeding both audio and video into an RTSP
3121 pipeline via appsrc.
3123 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
3125 * gst/rtsp-server/rtsp-client.c:
3126 client: Strip transport parts as whitespaces could be around commas
3127 https://bugzilla.gnome.org/show_bug.cgi?id=758428
3129 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
3131 * gst/rtsp-server/rtsp-stream.c:
3132 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
3133 Fix race when setting up source elements.
3134 Since we set the source element(s) to PLAYING state before hooking
3135 them up to the downstream funnel, it's possible for the source element
3136 to receive packets before we actually get to linking it to the funnel,
3137 in which case buffers would be pushed out on an unlinked pad, causing
3138 it to error out and stop receiving more data.
3139 We fix this by blocking the source's srcpad until we have linked it.
3140 https://bugzilla.gnome.org/show_bug.cgi?id=796160
3142 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
3144 * gst/rtsp-server/rtsp-stream.c:
3145 rtsp-stream: Fix mismatch between allowed and configured protocols
3146 https://bugzilla.gnome.org/show_bug.cgi?id=796679
3148 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
3150 * gst/rtsp-server/rtsp-stream.c:
3151 rtsp-stream: Emit a signal when the SRTP decoder is created
3152 https://bugzilla.gnome.org/show_bug.cgi?id=778080
3154 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
3156 * gst/rtsp-server/rtsp-stream.c:
3157 rtsp-stream: Don't require presence of sinks in _get_*_socket()
3158 Transport specific sink elements are added to the pipeline
3159 in PLAY request and sockets are already created in SETUP so
3160 it's actually wrong to require the presence of sinks in
3161 _get_*_socket() functions.
3162 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3164 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
3166 * gst/rtsp-server/rtsp-stream.c:
3167 rtsp-stream: Update transport for multicast clients as well
3168 If a multicast client requests different transport settings
3169 than the existing one make sure that this new transport
3170 configuruation is propagated to the multicast udp sink.
3171 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3173 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
3175 * gst/rtsp-server/rtsp-stream.c:
3176 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
3177 And not on unicast udp sinks
3178 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3180 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
3182 * gst/rtsp-server/rtsp-address-pool.c:
3183 * gst/rtsp-server/rtsp-auth.c:
3184 * gst/rtsp-server/rtsp-client.c:
3185 * gst/rtsp-server/rtsp-media-factory-uri.c:
3186 * gst/rtsp-server/rtsp-media-factory.c:
3187 * gst/rtsp-server/rtsp-media.c:
3188 * gst/rtsp-server/rtsp-mount-points.c:
3189 * gst/rtsp-server/rtsp-server.c:
3190 * gst/rtsp-server/rtsp-session-media.c:
3191 * gst/rtsp-server/rtsp-session-pool.c:
3192 * gst/rtsp-server/rtsp-session.c:
3193 * gst/rtsp-server/rtsp-stream-transport.c:
3194 * gst/rtsp-server/rtsp-stream.c:
3195 * gst/rtsp-server/rtsp-thread-pool.c:
3196 Update for g_type_class_add_private() deprecation in recent GLib
3198 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
3200 * gst/rtsp-server/rtsp-auth.c:
3201 * gst/rtsp-server/rtsp-media.c:
3202 * gst/rtsp-server/rtsp-sdp.c:
3203 * gst/rtsp-server/rtsp-stream.c:
3206 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
3208 * examples/Makefile.am:
3209 * examples/test-video-disconnect.c:
3210 examples: Add test-video-disconnect example
3211 Simple example which cuts off all clients 10 seconds
3212 after the first one connects.
3214 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3216 * docs/libs/gst-rtsp-server-sections.txt:
3217 * examples/test-auth-digest.c:
3218 * gst/rtsp-server/rtsp-auth.c:
3219 * gst/rtsp-server/rtsp-auth.h:
3220 rtsp-auth: Add support for parsing .htdigest files
3221 Passwords are usually not stored in clear text, but instead
3222 stored already hashed in a .htdigest file.
3223 Add support for parsing such files, add API to allow setting
3224 a custom realm in RTSPAuth, and update the digest example.
3225 https://bugzilla.gnome.org/show_bug.cgi?id=796637
3227 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
3229 * gst/rtsp-sink/gstrtspclientsink.c:
3230 * gst/rtsp-sink/gstrtspclientsink.h:
3231 rtspclientsink: fix waiting for multiple streams
3232 We were previously only ever waiting for a single stream to notify it's
3233 blocked status through GstRTSPStreamBlocking. Actually count streams to
3235 Fixes rtspclientsink sending SDP's without out some of the input
3237 https://bugzilla.gnome.org/show_bug.cgi?id=796624
3239 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3241 * docs/libs/gst-rtsp-server-sections.txt:
3242 docs: add missing auth methods
3244 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3246 * gst/rtsp-server/rtsp-stream.c:
3247 rtsp-stream: only create funnel if it didn't exist already.
3248 This precented using multiple protocols for the same stream.
3249 https://bugzilla.gnome.org/show_bug.cgi?id=796634
3251 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3253 * examples/meson.build:
3254 meson: build auth-digest example
3256 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
3258 * gst/rtsp-server/rtsp-client.c:
3259 * gst/rtsp-server/rtsp-media.c:
3260 * gst/rtsp-server/rtsp-sdp.c:
3261 * gst/rtsp-server/rtsp-session-media.c:
3262 * gst/rtsp-server/rtsp-stream-transport.c:
3263 Get payloader stats only for the sending streams
3264 Get/set payloader properties only for streams that actually
3265 contain a payloader element.
3266 https://bugzilla.gnome.org/show_bug.cgi?id=796523
3268 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
3270 * gst/rtsp-server/Makefile.am:
3271 Makefile: Don't hardcode libtool for g-i build
3272 Similar to the other commits in core/base/bad
3274 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
3276 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3277 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
3278 https://bugzilla.gnome.org/show_bug.cgi?id=796229
3280 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
3282 * gst/rtsp-sink/gstrtspclientsink.c:
3283 rtspclientsink: Don't deadlock in preroll on early close
3284 If the connection is closed very early, the flushing
3285 marker might not get set and rtspclientsink can get
3286 deadlocked waiting for preroll forever.
3287 https://bugzilla.gnome.org/show_bug.cgi?id=786961
3289 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3292 * meson_options.txt:
3293 meson: Update option names to omit disable_ and with- prefixes
3294 Also yield common options to the outer project (gst-build in our case)
3295 so that they don't have to be set manually.
3297 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3300 meson: use -Wl,-Bsymbolic-functions where supported
3301 Just like the autotools build.
3303 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3306 * tests/check/Makefile.am:
3307 configure: check for -good and -bad plugins only in uninstalled setup
3308 Avoids confusing configure messages looking or a -good .pc file
3310 Also use plugindir variables that common macros set while at it.
3311 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3313 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3315 * gst/rtsp-server/rtsp-client.c:
3316 rtsp-client: Fix session timeout
3317 When streaming data over TCP then is not the keep-alive
3318 functionality working.
3319 The reason is that the function do_send_data have changed
3320 to boolean but the code is still checking the received result
3321 from send_func with GST_RTSP_OK.
3322 The result is that a successful send_func will always lead to
3323 that do_send_data is returning false and the keep-alive will
3325 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3327 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3329 * docs/libs/gst-rtsp-server-sections.txt:
3330 * gst/rtsp-server/rtsp-media.c:
3331 * gst/rtsp-server/rtsp-sdp.c:
3332 * gst/rtsp-server/rtsp-stream.c:
3333 * gst/rtsp-server/rtsp-stream.h:
3334 * gst/rtsp-sink/gstrtspclientsink.c:
3335 * gst/rtsp-sink/gstrtspclientsink.h:
3336 Implement support for ULP Forward Error Correction
3337 In this initial commit, interface is only exposed for RECORD,
3338 further work will be needed in rtspsrc to support this for
3340 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3342 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3344 * gst/rtsp-server/rtsp-onvif-media.c:
3345 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3346 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3347 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3348 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3349 the opposite, just like the ONVIF standard.
3350 Let's follow those RFCs as we're doing RTSP here, and add a property at
3351 a later time if needed to switch to the SDP RFC behaviour.
3352 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3354 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3357 Automatic update of common submodule
3358 From 3fa2c9e to ed78bee
3360 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3362 * gst/rtsp-server/rtsp-client.c:
3363 * gst/rtsp-server/rtsp-media-factory.c:
3364 * gst/rtsp-server/rtsp-media.c:
3365 * gst/rtsp-server/rtsp-stream.c:
3366 * tests/check/gst/rtspclientsink.c:
3367 gst: Run everything through gst-indent again
3369 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3371 * gst/rtsp-server/rtsp-media.c:
3372 * tests/check/gst/media.c:
3373 rtsp-media: query the position on active streams if media is complete
3374 If the media is complete, i.e. one or more streams have been configured
3375 with sinks, then we want to query the position on those streams only.
3376 A query on an incomplete stream may return a position that originates from
3378 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3380 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3382 * gst/rtsp-sink/gstrtspclientsink.c:
3383 rtspclientsink: make sure not to use freed string
3384 Set transport string to NULL after freeing it, so that
3385 at worst we get a NULL pointer if constructing a new
3386 transport string fails (which shouldn't really fail here).
3387 Also check return value of that, just in case.
3390 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3392 * gst/rtsp-server/rtsp-client.c:
3393 rtsp-client: do not free string passed to take_header
3395 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3397 * gst/rtsp-server/rtsp-stream.c:
3398 rtsp-stream: do not take lock in request_aux_receiver
3399 Added it right before pushing the previous commit, it is
3400 incorrect and deadlocks because this function gets called
3401 from the join_bin thread, which already holds the lock,
3402 that's the reason why request_aux_sender didn't take the
3405 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3407 * docs/libs/gst-rtsp-server-sections.txt:
3408 * gst/rtsp-server/rtsp-media-factory.c:
3409 * gst/rtsp-server/rtsp-media-factory.h:
3410 * gst/rtsp-server/rtsp-media.c:
3411 * gst/rtsp-server/rtsp-media.h:
3412 * gst/rtsp-server/rtsp-stream.c:
3413 * gst/rtsp-server/rtsp-stream.h:
3414 rtsp-server: add API to enable retransmission requests
3415 "do-retransmission" was previously set when rtx-time != 0,
3416 which made no sense as do-retransmission is used to enable
3417 the sending of retransmission requests, where as rtx-time
3418 is used by the peer to enable storing of buffers in order
3419 to respond to retransmission requests.
3420 rtsp-media now also provides a callback for the
3421 request-aux-receiver signal.
3422 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3424 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3426 * gst/rtsp-sink/gstrtspclientsink.c:
3427 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3428 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3430 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3432 * gst/rtsp-sink/gstrtspclientsink.c:
3433 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3434 This in order to be able to decrypt the RTCP backchannel
3435 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3437 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3439 * gst/rtsp-server/rtsp-client.c:
3440 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3441 When sending back an encrypted RTCP back channel, it is useful
3442 for the client to know the encryption key.
3443 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3445 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3447 * gst/rtsp-server/rtsp-client.c:
3448 * gst/rtsp-server/rtsp-stream.c:
3449 * gst/rtsp-server/rtsp-stream.h:
3450 rtsp-stream: extract handle_keymgmt from rtsp-client
3451 rtspclientsink will also need to parse KeyMgmt headers
3452 sent by the server to decrypt the RTCP backchannel stream
3453 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3455 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3457 * gst/rtsp-sink/gstrtspclientsink.c:
3458 * tests/check/gst/rtspclientsink.c:
3459 rtspclientsink: Fix client ports for the RTCP backchannel
3460 This was broken since the work for delayed transport creation
3461 was merged: the creation of the transports string depends on
3462 calling stream_get_server_port, which only starts returning
3463 something meaningful after a call to stream_allocate_udp_sockets
3464 has been made, this function expects a transport that we parse
3465 from the transport string ...
3466 Significant refactoring is in order, but does not look entirely
3467 trivial, for now we put a band aid on and create a second transport
3468 string after the stream has been completed, to pass it in
3469 the request headers instead of the previous, incomplete one.
3470 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3472 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3474 * gst/rtsp-server/rtsp-client.c:
3475 rtsp-client:Error handling when equal http session cookie
3476 There are some clients that are sending same session cookie on random
3478 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3480 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3482 * gst/rtsp-server/rtsp-media-factory-uri.c:
3483 rtsp-media-factory-uri: Fix compilation with latest GLib
3484 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3485 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3486 data->factory = g_object_ref (factory);
3489 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3497 === release 1.14.0 ===
3499 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3505 * gst-rtsp-server.doap:
3509 === release 1.13.91 ===
3511 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3517 * gst-rtsp-server.doap:
3521 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3523 * gst/rtsp-server/Makefile.am:
3524 * gst/rtsp-server/meson.build:
3525 * gst/rtsp-server/rtsp-address-pool.h:
3526 * gst/rtsp-server/rtsp-auth.h:
3527 * gst/rtsp-server/rtsp-client.h:
3528 * gst/rtsp-server/rtsp-context.h:
3529 * gst/rtsp-server/rtsp-media-factory-uri.h:
3530 * gst/rtsp-server/rtsp-media-factory.h:
3531 * gst/rtsp-server/rtsp-media.h:
3532 * gst/rtsp-server/rtsp-mount-points.h:
3533 * gst/rtsp-server/rtsp-onvif-client.h:
3534 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3535 * gst/rtsp-server/rtsp-onvif-media.h:
3536 * gst/rtsp-server/rtsp-onvif-server.h:
3537 * gst/rtsp-server/rtsp-params.h:
3538 * gst/rtsp-server/rtsp-permissions.h:
3539 * gst/rtsp-server/rtsp-sdp.h:
3540 * gst/rtsp-server/rtsp-server-prelude.h:
3541 * gst/rtsp-server/rtsp-server.h:
3542 * gst/rtsp-server/rtsp-session-media.h:
3543 * gst/rtsp-server/rtsp-session-pool.h:
3544 * gst/rtsp-server/rtsp-session.h:
3545 * gst/rtsp-server/rtsp-stream-transport.h:
3546 * gst/rtsp-server/rtsp-stream.h:
3547 * gst/rtsp-server/rtsp-thread-pool.h:
3548 * gst/rtsp-server/rtsp-token.h:
3549 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3550 We need different export decorators for the different libs.
3551 For now no actual change though, just rename before the release,
3552 and add prelude headers to define the new decorator to GST_EXPORT.
3554 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3556 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3557 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3558 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3560 === release 1.13.90 ===
3562 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3568 * gst-rtsp-server.doap:
3572 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3574 * gst/rtsp-server/rtsp-media-factory.c:
3575 * gst/rtsp-server/rtsp-permissions.c:
3576 permissions: add Since tags and example for new API
3578 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3580 * docs/libs/gst-rtsp-server-sections.txt:
3581 * gst/rtsp-server/rtsp-media-factory.c:
3582 * gst/rtsp-server/rtsp-media-factory.h:
3583 * gst/rtsp-server/rtsp-permissions.c:
3584 * gst/rtsp-server/rtsp-permissions.h:
3585 * tests/check/gst/permissions.c:
3586 permissions: more bindings-friendly API
3587 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3589 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3592 meson: enable more warnings
3594 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3596 * gst/rtsp-server/rtsp-client.c:
3597 rtsp-client: Place netaddress meta on packets received via TCP
3598 This allows us to later map signals from rtpbin/rtpsource back to the
3599 corresponding stream transport, and allows to do keep-alive based on
3600 RTCP packets in case of TCP media transport.
3601 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3603 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3605 * gst/rtsp-sink/gstrtspclientsink.c:
3606 rtspclientsink: if OPEN failed, unqueue next command
3607 As READY_TO_PAUSED can no longer return async, the RECORD
3608 command will be queued before the OPEN command fails
3609 (for example in case the server could not be connected),
3610 and record then waits for ever.
3611 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3613 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3615 * gst/rtsp-sink/gstrtspclientsink.c:
3616 rtspclientsink: fix retrieval of custom payloader caps
3617 If a bin is passed as the custom payloader, the caps of
3618 its factory will be empty, the correct way to obtain the caps
3619 is to query its sinkpad.
3621 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3623 * gst/rtsp-sink/gstrtspclientsink.c:
3624 rtspclientsink: fix extra unref of custom payloader
3626 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3628 * gst/rtsp-sink/gstrtspclientsink.c:
3629 rspclientsink: fix recent code indentation
3631 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3633 * gst/rtsp-sink/gstrtspclientsink.c:
3634 rtspclientsink: add missing get_type prototype
3636 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3638 * gst/rtsp-sink/gstrtspclientsink.c:
3639 rtspclientsink: allow setting payloader as pad property
3640 This was a FIXME item, and can be quite useful, also
3641 allowing to specify payloader properties from the command
3642 line, which is always nice.
3643 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3645 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3647 * gst/rtsp-server/rtsp-media.c:
3648 rtsp-media: Replace g_print() log line
3649 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3651 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3653 * gst/rtsp-server/rtsp-media.c:
3654 * tests/check/gst/rtspclientsink.c:
3655 rtsp-media: fix RECORD getting stuck
3656 The test_record case was working because async=false had
3657 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3658 but that was incorrect, as it should not be needed.
3659 Removing async=false made the test fail as expected, this is
3660 fixed by not trying to preroll when preparing the media for
3661 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3662 and our peer will not start sending media until it has received
3663 a response to that request, and sent and received a response
3664 to RECORD as well, thus obviously preventing preroll.
3665 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3667 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3669 * gst/rtsp-server/rtsp-auth.c:
3670 rtsp-auth: fix set_tls_authentication_mode annotation
3672 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3674 * gst/rtsp-server/rtsp-onvif-media.c:
3675 rtp-server: remove redefined variable
3676 res is a boolean variable which is defined in the function scope and
3677 redefined, with no reason, in the loop scope. This patch removes the
3679 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3681 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3683 * gst/rtsp-server/rtsp-media.c:
3684 * gst/rtsp-server/rtsp-stream.c:
3685 * gst/rtsp-server/rtsp-stream.h:
3686 stream: Add functions for checking if stream is receiver or sender
3687 ...and replace all checks for RECORD in GstRTSPMedia which are really
3688 for "sender-only". This way the code becomes more generic and introducing
3689 support for onvif-backchannel later on will require no changes in
3692 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3694 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3695 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3696 onvif: Make requires_backchannel() public
3697 ...in order to let subclasses building the onvif part of the pipeline
3698 check whether backchannel shall be included or not.
3700 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3702 * gst/rtsp-server/rtsp-onvif-media.c:
3703 rtsp-server: Switch around sendonly/recvonly attributes
3704 They are wrong in the ONVIF streaming spec. The backchannel should be
3705 recvonly and the normal media should be sendonly: direction is always
3706 from the point of view of the SDP offerer (the server) according to
3709 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3711 * docs/libs/gst-rtsp-server-docs.sgml:
3712 * docs/libs/gst-rtsp-server-sections.txt:
3713 * examples/.gitignore:
3714 * examples/Makefile.am:
3715 * examples/test-onvif-backchannel.c:
3716 * gst/rtsp-server/Makefile.am:
3717 * gst/rtsp-server/rtsp-media.h:
3718 * gst/rtsp-server/rtsp-onvif-client.c:
3719 * gst/rtsp-server/rtsp-onvif-client.h:
3720 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3721 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3722 * gst/rtsp-server/rtsp-onvif-media.c:
3723 * gst/rtsp-server/rtsp-onvif-media.h:
3724 * gst/rtsp-server/rtsp-onvif-server.c:
3725 * gst/rtsp-server/rtsp-onvif-server.h:
3726 * gst/rtsp-server/rtsp-sdp.c:
3727 * gst/rtsp-server/rtsp-sdp.h:
3728 rtsp: Add support for ONVIF backchannel
3729 This adds a new RTSP server, client, media-factory and media subclass
3730 for handling the specifics of the backchannel. Ideally this later can be
3731 extended with other ONVIF specific features.
3733 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3735 * gst/rtsp-server/rtsp-media.c:
3736 rtsp-media: Add support for sending+receiving medias
3737 We need to add an appsrc/appsink in that case because otherwise the
3738 media bin will be a sink and a source for rtpbin, causing a pipeline
3740 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3742 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3748 === release 1.13.1 ===
3750 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3754 * gst-rtsp-server.doap:
3758 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3760 * gst/rtsp-server/rtsp-session-pool.c:
3761 session-pool: remove nullable return annotation
3762 create_watch can only return NULL from the API guards, no
3765 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3767 * gst/rtsp-server/rtsp-media-factory.c:
3768 * gst/rtsp-server/rtsp-media.c:
3769 set_clock functions: Add nullable annotations
3771 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3773 * gst/rtsp-server/rtsp-auth.c:
3774 * gst/rtsp-server/rtsp-client.c:
3775 * gst/rtsp-server/rtsp-media-factory.c:
3776 * gst/rtsp-server/rtsp-media.c:
3777 * gst/rtsp-server/rtsp-mount-points.c:
3778 * gst/rtsp-server/rtsp-server.c:
3779 * gst/rtsp-server/rtsp-session-media.c:
3780 * gst/rtsp-server/rtsp-session-pool.c:
3781 * gst/rtsp-server/rtsp-session.c:
3782 * gst/rtsp-server/rtsp-stream-transport.c:
3783 * gst/rtsp-server/rtsp-stream.c:
3784 * gst/rtsp-server/rtsp-thread-pool.c:
3785 All around: add annotations and API guards
3787 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3789 * tests/test-cleanup.c:
3790 test-cleanup: bind any port
3791 The meson test suite runs tests in parallel, trying to bind
3792 a single port made the test fail.
3794 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3797 meson: make version numbers ints and fix int/string comparison
3798 WARNING: Trying to compare values of different types (str, int).
3799 The result of this is undefined and will become a hard error
3800 in a future Meson release.
3802 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3804 * gst/rtsp-server/rtsp-context.c:
3805 gst_rtsp_context_get_current: add (skip) annotation
3806 The return value type is defined with G_DEFINE_POINTER_TYPE,
3807 and gi emits the following warning:
3808 Invalid non-constant return of bare structure or union; register as
3809 boxed type or (skip)
3811 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3813 * gst/rtsp-server/rtsp-client.c:
3814 rtsp-client: add type annotations
3815 gi doesn't seem to be able to figure out the type of the
3816 signal parameters when defined with G_DEFINE_POINTER_TYPE
3818 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3821 autotools: use -fno-strict-aliasing where supported
3822 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3824 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3827 meson: use -fno-strict-aliasing where supported
3828 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3830 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3832 * gst/rtsp-server/rtsp-mount-points.c:
3833 mount-points: bail out of loop again when matching mount points
3834 Previous patch led to us iterating the entire sequence. Bail out
3835 of the loop again if we have a match but are moving away from it.
3836 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3838 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3840 * tests/check/gst/mountpoints.c:
3841 tests: mountpoints: add more checks for mount point path matching
3842 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3844 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3846 * gst/rtsp-server/rtsp-mount-points.c:
3847 mount-points: fix matching of paths where there's also an entry with a common prefix
3848 e.g. with the following mount points
3852 _match() would not match /raw/video and /raw/snapshot correctly.
3853 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3855 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3857 * docs/libs/gst-rtsp-server-sections.txt:
3858 * gst/rtsp-server/rtsp-permissions.c:
3859 * gst/rtsp-server/rtsp-permissions.h:
3860 * tests/check/gst/permissions.c:
3861 permissions: add some new API to make this usable from bindings
3862 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3864 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3866 * gst/rtsp-server/rtsp-token.c:
3867 rtsp-token: annotate constructors for bindings
3868 This maps _new_empty() to _new(), which also makes RTSPToken()
3869 work properly now. Since this API wasn't usable from bindings
3870 before, this should hopefully be fine.
3871 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3873 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3875 * docs/libs/gst-rtsp-server-sections.txt:
3876 * gst/rtsp-server/rtsp-token.c:
3877 * gst/rtsp-server/rtsp-token.h:
3878 * tests/check/gst/token.c:
3879 rtsp-token: add some API to set fields from bindings
3880 The existing functions are all vararg-based and as such
3881 not usable from bindings.
3882 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3884 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3886 * tests/check/gst/rtspclientsink.c:
3887 * tests/check/gst/rtspserver.c:
3888 * tests/check/gst/sessionpool.c:
3889 * tests/check/gst/stream.c:
3890 tests: fix indentation
3893 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3895 * tests/check/gst/rtspserver.c:
3896 tests: rtspserver: fix another ref leak
3897 Even if this didn't show up in valgrind.
3899 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3901 * tests/check/gst/rtspclientsink.c:
3902 tests: rtspclientsink: fix leak
3904 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3906 * tests/check/gst/rtspserver.c:
3907 test: rtspserver: plug memory leak in test_no_session_timeout
3908 In test_no_session_timeout, unref the rtsp session object when the
3910 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3912 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3914 * gst/rtsp-sink/gstrtspclientsink.c:
3915 rtpsclientsink: Initialize and clear newly added mutex and cond
3916 While it *did* work, glib would automatically create new mutex and cond
3917 ... which never got freed
3919 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3921 * gst/rtsp-server/rtsp-stream.c:
3922 rtsp-stream: Set multicast TTL on the multicast sockets
3923 And not if we do unicast UDP.
3924 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3926 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3928 * gst/rtsp-server/rtsp-stream.c:
3929 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3930 In the multicast case (as in test-multicast, not test-multicast2), the
3931 address could be allocated/reserved (and thus set) already without
3932 allocating the actual socket. We need to allocate the socket here still
3933 instead of just claiming that it was already allocated.
3934 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3936 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3938 * gst/rtsp-sink/gstrtspclientsink.c:
3939 * gst/rtsp-sink/gstrtspclientsink.h:
3940 rtspclientsink: Use the new rtsp-stream API
3941 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3943 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3945 * gst/rtsp-sink/gstrtspclientsink.c:
3946 * gst/rtsp-sink/gstrtspclientsink.h:
3947 rtspclientsink: Wait until OPEN has been scheduled
3948 Make sure that the sink thread has started opening connection
3949 to the server before continuing.
3950 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3952 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3955 Automatic update of common submodule
3956 From e8c7a71 to 3fa2c9e
3958 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3960 * gst/rtsp-server/rtsp-media.c:
3961 * gst/rtsp-server/rtsp-session-media.c:
3962 * gst/rtsp-server/rtsp-stream.c:
3963 rtsp-server: Minor doc fixes
3966 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3969 * tests/Makefile.am:
3970 tests: disable all tests when --disable-tests is used
3971 Move conditional subdir include into top level.
3972 Based on patch by: Joel Holdsworth
3973 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3975 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3978 * meson_options.txt:
3979 * tests/meson.build:
3980 meson: build more tests and add options to disable tests and examples
3982 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3984 * gst/rtsp-server/rtsp-session.c:
3985 Fix build when -Werror=deprecated-declarations is on
3986 As gst_rtsp_session_next_timeout is deprecated.
3988 ../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]
3989 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3991 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3992 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3993 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3996 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3999 Automatic update of common submodule
4000 From 3f4aa96 to e8c7a71
4002 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4004 * tests/check/gst/media.c:
4005 check/media: Add seekability test case: not all streams are active
4006 Media contains two streams but only one is complete and prepared
4008 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4010 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4012 * gst/rtsp-server/rtsp-stream.c:
4013 rtsp-stream: Do not reset 'blocking' if stream is already blocked
4014 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4016 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4018 * gst/rtsp-server/rtsp-media.c:
4019 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
4020 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4022 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
4025 meson: remove vs_module_defs_dir variable which is no longer needed
4027 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
4029 * gst/rtsp-server/rtsp-session.h:
4032 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
4035 * gst/rtsp-server/meson.build:
4037 * win32/common/libgstrtspserver.def:
4038 win32: remove .def file with exports
4039 They're no longer needed, symbol exporting is now explicit
4040 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
4042 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4045 autotools: stop controlling symbol visibility with -export-symbols-regex
4046 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
4047 This should result in consistent behaviour for the autotools and
4050 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4052 * gst/rtsp-server/rtsp-media.h:
4053 * gst/rtsp-server/rtsp-server.h:
4054 * gst/rtsp-server/rtsp-session.c:
4055 * gst/rtsp-server/rtsp-session.h:
4056 rtsp-server: add missing GST_EXPORT and export deprecated funcs
4058 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
4060 * tests/check/gst/media.c:
4061 check: Add seekability testing on medias
4062 Make sure that once GstRTSPMedia are prepared they returned
4063 the expected seekability results
4064 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4066 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
4068 * docs/libs/gst-rtsp-server-sections.txt:
4069 * gst/rtsp-server/rtsp-media.c:
4070 * gst/rtsp-server/rtsp-stream.c:
4071 * gst/rtsp-server/rtsp-stream.h:
4072 * win32/common/libgstrtspserver.def:
4073 rtsp-media: Enable seeking query before pipeline is complete
4074 SDP are now provided *before* the pipeline is fully complete. In order
4075 to know whether a media is seekable or not therefore requires asking
4076 the invididual streams.
4077 API: gst_rtsp_stream_seekable
4078 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4080 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
4082 * gst/rtsp-server/rtsp-media.c:
4083 rtsp-media: Fix handling in default_unsuspend()
4084 Handle the case when streams are not blocked and media
4085 is suspended from PAUSED.
4086 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
4087 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4089 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
4091 * tests/check/gst/media.c:
4092 check/media: Fix thread pool leak.
4093 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
4094 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4096 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
4098 * gst/rtsp-server/rtsp-media.c:
4099 rtsp-media: Removed fakesink elements
4100 There is not need of adding fakesink elements to the media
4101 pipeline in the dynamic-payloader case.
4102 The media pipeline itself is dynamically updated with
4103 the receiver and sender parts that are based on the client
4104 transport information known after SETUP has been received.
4105 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
4106 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4108 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
4110 * gst/rtsp-server/rtsp-media.c:
4111 rtsp-media: Corrected ASYNC_DONE handling
4112 Media is complete when all the transport based parts are
4113 added to the media pipeline. At this point ASYNC_DONE is
4114 posted by the media pipeline and media is ready to enter
4116 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
4117 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4119 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
4121 * tests/check/gst/media.c:
4122 check/media: Check that prepared media can provide a SDP
4123 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
4125 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
4127 * gst/rtsp-server/rtsp-client.c:
4128 rtsp-client: Don't leak addr
4131 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
4133 * gst/rtsp-server/rtsp-client.c:
4134 * gst/rtsp-server/rtsp-session-media.c:
4135 * gst/rtsp-server/rtsp-stream.c:
4138 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
4140 * gst/rtsp-server/rtsp-media.c:
4141 rtsp-media: Don't unblock with remaining dynamic payloaders
4142 If we still have some dynamic paylaoders which haven't posted
4143 no-more-pads yet, don't go to PREPARED if one of the streams
4145 The risk was that we would end up not exposing/using all specified
4147 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
4148 then it will take a bit more time to start. But only if those 3
4149 conditions are present.
4150 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4152 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
4154 * gst/rtsp-server/rtsp-media.c:
4157 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
4159 * gst/rtsp-server/rtsp-media.c:
4160 rtsp-media: Don't set float on a gint64 variable
4161 Just use 0. Fixes 'undefined' behaviour from clang
4163 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
4165 * gst/rtsp-server/rtsp-media.c:
4166 rtsp-media: Fix previous commit
4167 We only want to count dynamic payloaders
4169 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
4171 * gst/rtsp-server/rtsp-media.c:
4172 * tests/check/gst/media.c:
4173 rtsp-media: Handle multiple dynamic elements
4174 If we have more than one dynamic payloader in the pipeline, we need
4175 to wait until the *last* one emits 'no-more-pads' before switching
4177 Failure to do so would result in a race where some of the streams
4178 wouldn't properly be prepared
4179 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4181 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4183 * win32/common/libgstrtspserver.def:
4184 win32: Fix exported symbols list
4186 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
4188 * gst/rtsp-server/rtsp-stream.c:
4189 rtsp-stream: Only update the RTP udpsink if it actually exists
4190 For send-only streams it does not exist, but the RTCP udpsink might.
4192 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
4194 * win32/common/libgstrtspserver.def:
4195 win32: Update exports
4197 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
4199 * gst/rtsp-server/rtsp-media.c:
4200 * gst/rtsp-server/rtsp-stream.c:
4201 * gst/rtsp-server/rtsp-stream.h:
4202 rtsp-media: seek on media pipelines that are complete
4203 Make sure that a seek is performed on pipelines that
4204 contain at least one sink element.
4205 Change-Id: Icf398e10add3191d104b1289de612412da326819
4206 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4208 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
4210 * gst/rtsp-server/rtsp-client.c:
4211 * gst/rtsp-server/rtsp-media.c:
4212 * gst/rtsp-server/rtsp-media.h:
4213 * gst/rtsp-server/rtsp-stream.c:
4214 * gst/rtsp-server/rtsp-stream.h:
4215 * tests/check/gst/client.c:
4216 * tests/check/gst/media.c:
4217 * tests/check/gst/rtspserver.c:
4218 * tests/check/gst/stream.c:
4219 Dynamically reconfigure pipeline in PLAY based on transports
4220 The initial pipeline does not contain specific transport
4221 elements. The receiver and the sender parts are added
4223 If the media is shared, the streams are dynamically
4224 reconfigured after each PLAY.
4225 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4227 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
4229 * gst/rtsp-server/rtsp-stream.c:
4230 rtsp-stream: obtain stream position from pad
4231 If no sinks have been added yet, obtain the current and
4232 the stop position of the stream from the send_src pad.
4233 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
4234 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4236 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
4238 * gst/rtsp-server/rtsp-session-media.c:
4239 * gst/rtsp-server/rtsp-session-media.h:
4240 rtsp-session-media: add function to get a list of transports
4241 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
4242 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4244 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
4246 * gst/rtsp-server/rtsp-stream.c:
4247 * gst/rtsp-server/rtsp-stream.h:
4248 rtsp-stream: add functions to get rtp and rtcp multicast sockets
4249 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
4250 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4252 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
4254 * gst/rtsp-server/rtsp-stream.c:
4255 stream: set async=sync=false only for RTCP appsink
4256 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
4257 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4259 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
4261 * gst/rtsp-server/rtsp-media.c:
4262 rtsp-media: return minimum value in query position case
4263 The minimum position should be returned as we are interested
4264 in the whole interval.
4265 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
4266 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4268 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
4270 * gst/rtsp-server/rtsp-session.c:
4271 * tests/check/gst/rtspserver.c:
4272 rtsp-session: Handle the case when timeout=0
4273 According to the documentation, a timeout of value 0 means
4274 that the session never timeouts. This adds handling of that.
4275 If timeout=0 we just return with a -1 from
4276 gst_rtsp_session_next_timeout_usec ().
4277 https://bugzilla.gnome.org/show_bug.cgi?id=785058
4279 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4281 * gst/rtsp-sink/gstrtspclientsink.c:
4282 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
4283 https://bugzilla.gnome.org/show_bug.cgi?id=785024
4285 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
4287 * docs/libs/gst-rtsp-server-sections.txt:
4288 * gst/rtsp-server/rtsp-media-factory.c:
4289 docs: add media factory transport mode accessors
4290 and fix the documentation for the return value of the getter
4292 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
4294 * gst/rtsp-server/rtsp-client.c:
4295 rtsp-client: unref 'pipelined_requests' in finalize
4296 The hash table priv->pipelined_requests is not unref:ed in the
4297 finalize funktion. Make sure it is.
4298 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4300 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4302 * gst/rtsp-server/rtsp-media.c:
4303 rtsp-media: Initialize scalar variable
4306 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4308 * win32/common/libgstrtspserver.def:
4309 win32: Update export file
4311 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4313 * gst/rtsp-server/rtsp-client.c:
4314 * gst/rtsp-server/rtsp-media.c:
4315 * gst/rtsp-server/rtsp-media.h:
4316 Start support for RTSP 2.0
4317 This adds basic support for new 2.0 features, though the protocol is
4318 subposdely backward incompatible, most semantics are the sames.
4321 * version negotiation
4322 * pipelined requests support
4323 * Media-Properties support
4324 * Accept-Ranges support
4326 * gst_rtsp_media_seekable
4327 The RTSP methods that have been removed when using 2.0 now return
4329 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4331 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4333 * gst/rtsp-server/rtsp-stream.c:
4334 stream: Use stream duration as stream-stop if segment was not configured with a stop
4335 Allowing client to know stream duration when no seeking happened.
4336 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4338 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4340 * gst/rtsp-server/rtsp-media-factory.c:
4341 rtsp-media-factory: Don't cache any media if NULL was returned as key
4342 The docs already mentioned this, but we actually stored it in the hash
4343 table with key==NULL and leaked its reference forever.
4345 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4347 * gst/rtsp-sink/gstrtspclientsink.c:
4348 * gst/rtsp-sink/gstrtspclientsink.h:
4349 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4350 This is a simple port of:
4351 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4352 * c438545dc9e2f14f657bc0ef261fff726449867b
4353 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4354 in gst-plugins-good.
4356 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4358 * gst/rtsp-server/rtsp-sdp.c:
4359 sdp: fix Memory leak in error case
4360 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4362 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4364 * pkgconfig/meson.build:
4365 meson: don't install -uninstalled.pc file
4366 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4368 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4371 Automatic update of common submodule
4372 From 48a5d85 to 3f4aa96
4374 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4376 * gst/rtsp-server/rtsp-client.c:
4377 rtsp-client: Fix typo in debug message
4379 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4382 meson: hide symbols by default unless explicitly exported
4384 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4386 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4387 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4388 Fixes meson warning about undefined @srcdir@.
4390 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4392 * tests/meson.build:
4393 meson: skip tests on windows for now
4394 As we do in the other modules. As libgstcheck is currently not
4395 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4396 the subproject 'gstreamer' does not exist"" Meson error.
4398 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4400 * gst/rtsp-server/rtsp-stream.c:
4401 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4402 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4403 multiudpsink's last-sample always comes from the payloader. Which
4404 is wrong if auxiliary streams are multiplexed in the same stream.
4405 So check the buffer's ssrc against the caps'ssrc before to use its
4406 seqnum. If not the same ssrc just use the payloader as done prior
4407 the commit above or when there is no last-sample yet.
4408 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4410 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4413 meson: Allow using glib as a subproject
4415 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4418 meson: fix with-package-name option
4419 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4421 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4424 Distribute meson_options.txt
4426 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4429 And config.h.meson is no longer dist either
4431 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4435 meson: config.h.meson is no longer needed
4437 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4439 * tests/check/meson.build:
4440 * tests/meson.build:
4441 meson: Fix building tests and activate them again
4443 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4445 * tests/check/meson.build:
4446 meson: Do not use path separator in test names
4447 Avoiding warnings like:
4448 WARNING: Target "elements/audioamplify" has a path separator in its name.
4450 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4453 * meson_options.txt:
4454 meson: add options to set package name and origin
4455 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4457 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4459 * gst/rtsp-server/rtsp-address-pool.h:
4460 * gst/rtsp-server/rtsp-auth.h:
4461 * gst/rtsp-server/rtsp-client.h:
4462 * gst/rtsp-server/rtsp-context.h:
4463 * gst/rtsp-server/rtsp-media-factory-uri.h:
4464 * gst/rtsp-server/rtsp-media-factory.h:
4465 * gst/rtsp-server/rtsp-media.h:
4466 * gst/rtsp-server/rtsp-mount-points.h:
4467 * gst/rtsp-server/rtsp-params.h:
4468 * gst/rtsp-server/rtsp-permissions.h:
4469 * gst/rtsp-server/rtsp-sdp.h:
4470 * gst/rtsp-server/rtsp-server.h:
4471 * gst/rtsp-server/rtsp-session-media.h:
4472 * gst/rtsp-server/rtsp-session-pool.h:
4473 * gst/rtsp-server/rtsp-session.h:
4474 * gst/rtsp-server/rtsp-stream-transport.h:
4475 * gst/rtsp-server/rtsp-stream.h:
4476 * gst/rtsp-server/rtsp-thread-pool.h:
4477 * gst/rtsp-server/rtsp-token.h:
4478 Mark symbols explicitly for export with GST_EXPORT
4480 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4483 * gst/rtsp-sink/Makefile.am:
4484 Remove plugin specific static build option
4485 Static and dynamic plugins now have the same interface. The standard
4486 --enable-static/--enable-shared toggle are sufficient.
4488 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4494 === release 1.12.0 ===
4496 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4502 * gst-rtsp-server.doap:
4506 === release 1.11.91 ===
4508 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4514 * gst-rtsp-server.doap:
4518 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4521 Automatic update of common submodule
4522 From 60aeef6 to 48a5d85
4524 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4526 * gst/rtsp-server/rtsp-media-factory.c:
4527 * gst/rtsp-server/rtsp-media.c:
4528 * gst/rtsp-server/rtsp-session.c:
4529 * gst/rtsp-server/rtsp-stream.c:
4530 gi: Fix some annotations and docstrings
4532 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4534 * gst/rtsp-server/meson.build:
4536 * meson_options.txt:
4539 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4543 Automatic update of common submodule
4544 From 39ac2f5 to 60aeef6
4546 === release 1.11.90 ===
4548 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4554 * gst-rtsp-server.doap:
4558 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4560 * examples/test-launch.c:
4561 examples: make test-launch pipeline shared by default as well
4563 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4565 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4566 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4567 Just the build dir is not going to work for srcdir!=builddir.
4569 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4572 meson: Update version
4574 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4579 === release 1.11.2 ===
4581 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4587 * gst-rtsp-server.doap:
4590 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4593 meson: dist meson build files
4594 Ship meson build files in tarballs, so people who use tarballs
4595 in their builds can start playing with meson already.
4597 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4599 * examples/test-record.c:
4600 examples/test-record: Add extra line to initial printout
4601 Add an example line of how to deliver a stream to the
4604 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4606 * gst/rtsp-server/rtsp-client.c:
4607 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4608 If there is no Content-Length header, no body would be allocated and the
4609 '\0' would also not be appended to the body.
4611 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4613 * gst/rtsp-server/rtsp-client.c:
4614 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4615 While they logically have 0 bytes length, GstRTSPConnection is appending
4616 a '\0' to everything making the size be 1 instead.
4618 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4623 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4625 * gst/rtsp-server/rtsp-session.c:
4626 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4627 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4630 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4635 === release 1.11.1 ===
4637 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4643 * gst-rtsp-server.doap:
4644 * win32/common/libgstrtspserver.def:
4647 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4649 * gst/rtsp-server/rtsp-stream.c:
4650 rtsp-stream: corrected if-statement in _get_server_port()
4651 This bug was accidentally introduced while fixing a segfault
4652 in _get_server_port() function.
4653 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4655 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4657 * gst/rtsp-server/rtsp-stream.c:
4658 * tests/check/gst/stream.c:
4659 rtsp-stream: fixed segmenation fault in _get_server_port()
4660 Calling function gst_rtsp_stream_get_server_port() results in
4661 segmenation fault in the RTP/RTSP/TCP case.
4662 Port that the server will use to receive RTCP makes only
4663 sense in the UDP case, however the function should handle
4664 the TCP case in a nicer way.
4665 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4667 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4669 * gst/rtsp-server/rtsp-media-factory.c:
4670 dosc: Fix a little typo
4671 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4673 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4675 * pkgconfig/Makefile.am:
4676 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4677 * pkgconfig/meson.build:
4678 meson: generate pkg-config -uninstalled pc files
4679 Generating those files is useful for users building the GStreamer stack
4680 using meson and having to link it to another project which is still
4681 using the autotools.
4682 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4684 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4686 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4687 pkgconfig: fix -uninstalled pc file
4688 pcfiledir was never defined so the paths were wrong.
4689 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4691 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4693 * gst/rtsp-server/rtsp-stream.c:
4694 * tests/check/gst/rtspserver.c:
4695 rtsp-stream: Fixed TCP transport case
4696 Make sure that the appsink element is actually added to
4697 the bin before trying to link it with the elements in it.
4698 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4700 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4706 Remove generated .spec file
4707 Likely extremely bitrotten, and we should not ship this anyway.
4709 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4712 Automatic update of common submodule
4713 From f980fd9 to 39ac2f5
4715 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4717 * gst/rtsp-server/rtsp-media.c:
4718 media: Fix pt map caps
4719 Since decryption is handled within rtpbin, all outcoming stream
4720 caps will be application/x-rtp (i.e. regular rtp)
4721 Fixes RECORD with SRTP streams
4723 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4725 * gst/rtsp-server/rtsp-media-factory.c:
4726 media-factory: Create media objects with the proper transport mode
4727 The function called immediately afterwards (collect_streams()) will
4728 need it to work properly
4730 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4732 * gst/rtsp-server/rtsp-auth.c:
4733 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4735 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4737 * gst/rtsp-server/rtsp-media-factory.c:
4738 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4739 We're going to put a pipeline into a pipeline otherwise, which is not
4742 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4744 * gst/rtsp-server/rtsp-media.c:
4745 media: Fix race condition around finish_unprepare() if called multiple time
4746 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4748 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4750 * gst/rtsp-sink/gstrtspclientsink.c:
4751 rtspclientsink: Don't leave stale pointer after unref
4752 Fix a warning on shutdown - don't keep a pointer to an
4753 alread-unreffed object.
4755 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4758 common: use https protocol for common submodule
4759 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4761 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4763 * gst/rtsp-server/rtsp-stream.c:
4764 stream: block the output of rtpbin instead of the source pipeline
4765 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4766 detection of the srtp rollover counter to add to the SDP.
4767 Unfortunately, it was incomplete for live pipelines where the logic
4768 blocks the source bin before creating the SDP and thus would never have
4769 the necessary informaiton to create a correct SDP with srtp encryption.
4770 Move the pad blocks to rtpbin's output pads instead so that the
4771 necessary information can be created before we need the information for
4773 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4775 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4777 * gst/rtsp-server/rtsp-client.c:
4778 rtsp-client: add IDLE timeout, before session exists
4779 The RTSP server will not timeout an idle RTSP connection
4780 (note this is different from doing timeout on a RTSP
4782 At least for Apache this is a problem when running RTSP over
4783 HTTPS since it uses one of the threads (there is a rather
4784 limited number) that are available for handling requests.
4785 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4787 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4792 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4794 * gst/rtsp-server/rtsp-stream.c:
4795 rtsp-stream: Set close-socket FALSE on UDP src:es
4796 With this RTSP server can use the sockets independent on the udpsrc
4798 When the udp src is finalized it will unref socket and when g_socket
4799 is finalized the socket will be closed.
4800 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4802 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4804 * gst/rtsp-sink/gstrtspclientsink.c:
4805 rtspclientsink: Move to new helper function to parse authentication responses
4806 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4808 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4810 * examples/Makefile.am:
4811 * examples/test-auth-digest.c:
4812 * gst/rtsp-server/rtsp-auth.c:
4813 * gst/rtsp-server/rtsp-auth.h:
4814 * win32/common/libgstrtspserver.def:
4815 rtsp-auth: Add support for Digest authentication
4816 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4818 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4821 * gst/rtsp-server/meson.build:
4823 * tests/check/meson.build:
4825 * win32/common/libgstrtspserver.def:
4826 Enable building with MSVC
4827 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4829 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4832 meson: gstreamer gst_check_dep does not exist on windows
4834 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4836 * gst/rtsp-server/rtsp-client.c:
4837 client: update do_send_message to match type GstRTSPClientSendFunc
4838 This type mismatch fails building with MSVC
4839 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4841 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4843 * gst/rtsp-server/rtsp-sdp.c:
4844 rtsp-sdp: Fix indentation
4846 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4848 * gst/rtsp-server/rtsp-media.c:
4849 rtsp-media: Only signal "new-state" if the state has actually changed
4850 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4852 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4854 * gst/rtsp-server/rtsp-client.c:
4855 * gst/rtsp-server/rtsp-client.h:
4856 client: emit signal in the beginning of each rtsp request
4857 These signals let the application validate the requests, configure the
4858 media/stream in a certain way and also generate error status code in
4859 case of error or bad request.
4860 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4862 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4865 meson: update version
4867 === release 1.11.0 ===
4869 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4874 === release 1.10.0 ===
4876 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4882 * gst-rtsp-server.doap:
4885 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4887 * tests/check/gst/rtspserver.c:
4888 * tests/check/gst/stream.c:
4889 tests: try to avoid using the same ports in different tests
4890 Causes problems with client multicast tests otherwise if
4891 tests are run in parallel.
4892 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4894 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4896 * tests/check/gst/client.c:
4897 tests: client: use fail_unless_equals_foo() for better failure reporting
4899 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4901 * gst/rtsp-server/rtsp-client.c:
4902 rtsp-client: Session filter in unwatch session
4903 Call session filter with filter_session_media as paramer in
4904 client_unwatch_session if using drop_backlog = FALSE.
4905 In client_unwatch_session its allowed to grow the watchs backlog.
4906 If using drop_backlog = FALSE and the backlog is full it will cause
4907 a deadlock when setting session media state to NULL
4908 if the backlog is not allowed to grow.
4909 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4911 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4914 meson: add fallbacks for gst modules
4917 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4919 * gst/rtsp-server/rtsp-client.c:
4920 rtsp-client: Fix factory leaking in find_media() in error cases
4921 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4923 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4925 * gst/rtsp-server/rtsp-stream.c:
4926 stream: Fix randomly missing streams from SDP with dynamic elements
4927 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4928 "pad-added" signal. In that case priv->srcpad could already have its caps,
4929 and they'll be sent to priv->send_src[0] pad. That means that when it
4930 connects "notify::caps" signal, that pad could already have received its
4931 caps and the signal won't be emitted anymore.
4932 In that case priv->caps stay to NULL and when building the SDP that stream
4933 gets ignored. Leading to missing video or audio when playing in client side.
4934 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4936 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4939 meson: update version
4941 === release 1.9.90 ===
4943 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4949 * gst-rtsp-server.doap:
4952 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4954 * gst/rtsp-server/rtsp-media-factory.c:
4955 * gst/rtsp-server/rtsp-media.c:
4956 * gst/rtsp-server/rtsp-stream.c:
4957 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4958 To prevent any possibly confusion with IPs or anything else.
4959 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4961 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4963 * gst/rtsp-server/rtsp-media-factory.c:
4964 * gst/rtsp-server/rtsp-media.c:
4965 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4966 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4968 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4971 configure: Depend on gstreamer 1.9.2.1
4973 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4977 Automatic update of common submodule
4978 From b18d820 to f980fd9
4980 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4984 Automatic update of common submodule
4985 From 6f2d209 to b18d820
4987 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4989 * gst/rtsp-server/rtsp-stream.c:
4990 rtsp-stream: Remove unused _locked() variant of a function
4991 It was added during refactoring.
4993 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4995 * gst/rtsp-server/rtsp-stream.c:
4996 stream: cosmetic cleanup
4997 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4999 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5001 * gst/rtsp-server/rtsp-stream.c:
5002 stream: Compare IP addresses case insensitive in more places
5003 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5005 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5008 * gst/rtsp-server/rtsp-stream.c:
5009 stream: Fix leaked joined_bin
5010 There is no need to keep a strong ref on it, and _leave_bin() was
5011 setting it to NULL before calling g_clear_object() so it was leaked.
5012 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5014 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5016 * gst/rtsp-server/rtsp-stream.c:
5017 rtsp-stream: Compare IP address strings case insensitive
5018 Otherwise IPv6 addresses might fail this comparision.
5020 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
5022 * gst/rtsp-server/rtsp-stream.c:
5023 rtsp-stream: Bind multicast sockets to ANY as before
5024 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
5026 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
5028 * gst/rtsp-server/rtsp-session.c:
5029 rtsp-session: Fix segfault when doing keep-alive after removing the session
5030 If keep-alive happens after removing the session but before finalizing the
5031 stream transport, we would segfault.
5032 https://bugzilla.gnome.org/show_bug.cgi?id=750544
5034 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
5036 * gst/rtsp-server/rtsp-stream.c:
5037 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
5038 Adding them later will cause deadlocks due to
5039 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
5040 2) adding the multicast sink
5041 3) waiting for it to get data to preroll again
5042 3) never happens because the queues after the tee are full.
5044 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
5046 * gst/rtsp-server/rtsp-stream.c:
5047 rtsp-stream: Fix up various multicast related issues
5049 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
5051 * tests/check/gst/stream.c:
5052 tests: Fix compilation
5054 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5056 * gst/rtsp-server/rtsp-client.c:
5057 * gst/rtsp-server/rtsp-stream.c:
5058 * tests/check/gst/stream.c:
5059 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
5060 This is basically reverting changes introduced in commit f62a9a7,
5061 because it was introducing various regressions:
5062 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
5063 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
5064 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
5065 - If a mcast client connects, it creates a new socket in SETUP to try to respect
5066 the destination/port given by the client in the transport, and overrides the
5067 socket already set on the udpsink element. That means that if we already had a
5068 client connected, the source address on the udp packets it receives suddenly
5070 - If a 2nd mcast client connects, the destination/port in its transport is
5071 ignored but its transport wasn't updated.
5072 What this patch does:
5073 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
5074 - Always have a tee+queue when udp is enabled. This could be optimized
5075 again in a later patch, but is more complicated. If no unicast clients
5076 connects then those elements are useless, this could be also optimized
5078 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
5079 seperated from those for unicast clients. Since we already support only
5080 one mcast address, we also create only one set of elements.
5081 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5083 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5085 * gst/rtsp-server/rtsp-stream.c:
5086 stream: factor our plug_src function
5087 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5089 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5091 * gst/rtsp-server/rtsp-stream.c:
5092 stream: factor out plug_sink function
5093 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5095 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5097 * gst/rtsp-server/rtsp-stream.c:
5098 stream: small documentation clarification
5099 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5101 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5103 * gst/rtsp-server/rtsp-stream.c:
5104 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
5105 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5107 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5109 * gst/rtsp-server/rtsp-stream.c:
5110 stream: Keep a ref on joined bin
5111 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5113 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5115 * gst/rtsp-server/rtsp-stream.c:
5116 stream: code cleanup
5117 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5119 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5121 * gst/rtsp-server/rtsp-stream.c:
5122 stream: small fix in error code path
5123 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5125 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5127 * gst/rtsp-server/rtsp-stream.c:
5128 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
5129 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
5130 but keeps unit tests.
5131 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5133 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
5138 === release 1.9.2 ===
5140 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
5146 * gst-rtsp-server.doap:
5149 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
5152 * examples/meson.build:
5154 * gst/rtsp-server/meson.build:
5155 * gst/rtsp-sink/meson.build:
5157 * pkgconfig/meson.build:
5158 * tests/check/meson.build:
5159 * tests/meson.build:
5160 Add support for Meson as alternative/parallel build system
5161 https://github.com/mesonbuild/meson
5163 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
5166 * tests/check/Makefile.am:
5167 build: silence error about pthread for 'make check' in osx
5168 Fixes "clang: error: argument unused during compilation: '-pthread'"
5170 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
5172 * gst/rtsp-server/rtsp-client.c:
5173 rtsp-client: Fix leaking of media in error cases
5174 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
5175 and myself to make the media refcounting a bit easier to follow.
5176 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5178 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
5180 * gst/rtsp-server/rtsp-client.c:
5181 rtsp-client: Fix leaking of session in error cases
5182 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5184 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
5187 Automatic update of common submodule
5188 From f363b32 to f49c55e
5190 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
5195 === release 1.9.1 ===
5197 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5203 * gst-rtsp-server.doap:
5206 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
5209 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
5210 https://bugzilla.gnome.org/show_bug.cgi?id=767463
5212 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5215 Automatic update of common submodule
5216 From ac2f647 to f363b32
5218 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5220 * gst/rtsp-server/rtsp-sdp.c:
5221 * gst/rtsp-server/rtsp-sdp.h:
5222 * gst/rtsp-server/rtsp-stream.c:
5223 * gst/rtsp-server/rtsp-stream.h:
5224 sdp: add rollover counters for all sender SSRC
5225 We add different crypto sessions in MIKEY, one for each sender
5226 SSRC. Currently, all of them will have the same security policy, 0.
5227 The rollover counters are obtained from the srtpenc element using the
5229 https://bugzilla.gnome.org/show_bug.cgi?id=730539
5231 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5233 * gst/rtsp-server/rtsp-media-factory.h:
5234 * gst/rtsp-server/rtsp-server.h:
5235 docs: fix some typos
5237 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
5239 * gst/rtsp-server/Makefile.am:
5240 g-i: pass compiler env to g-ir-scanner
5241 It's what introspection.mak does as well. Should
5242 fix spurious build failures on gnome-continuous
5243 (caused by g-ir-scanner getting compiler details
5244 via python which is broken in some environments
5245 so passing the compiler details bypasses that).
5247 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
5249 * gst/rtsp-server/rtsp-session.c:
5250 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
5251 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
5252 https://bugzilla.gnome.org/show_bug.cgi?id=766619
5254 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
5256 * gst/rtsp-sink/gstrtspclientsink.c:
5257 rtspclientsink: Check return value of sscanf
5258 And just make sure we always have 0/0 if we have an error
5261 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
5263 * gst/rtsp-server/rtsp-stream.c:
5264 * tests/check/gst/rtspserver.c:
5265 * tests/check/gst/stream.c:
5266 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
5267 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
5268 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
5269 - Create unit test for shared media.
5270 https://bugzilla.gnome.org/show_bug.cgi?id=764744
5272 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5274 * gst/rtsp-server/rtsp-stream.c:
5275 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
5276 For IPv6 addresses, binding to a multicast group does not work on Linux
5277 either. Always bind to ANY and then later join the multicast group.
5278 https://bugzilla.gnome.org/show_bug.cgi?id=764679
5280 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
5283 Automatic update of common submodule
5284 From 6f2d209 to ac2f647
5286 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
5288 * gst/rtsp-server/rtsp-thread-pool.c:
5289 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
5290 Clarified why it is necessary to add source information to
5291 GstRTSPThreadImpl. See the reported bug in GLib:
5292 https://bugzilla.gnome.org/show_bug.cgi?id=720186
5293 for more information.
5294 https://bugzilla.gnome.org/show_bug.cgi?id=761702
5296 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
5298 * examples/Makefile.am:
5299 examples: Clean up CFLAGS/LDADD even more
5300 The internal .la should come first and is part of LDADD, as is
5303 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5305 * examples/Makefile.am:
5306 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5308 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5310 * gst/rtsp-server/Makefile.am:
5311 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5313 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5315 * gst/rtsp-server/rtsp-client.c:
5316 * gst/rtsp-server/rtsp-media-factory.c:
5317 * gst/rtsp-server/rtsp-media-factory.h:
5318 * gst/rtsp-server/rtsp-media.c:
5319 * gst/rtsp-server/rtsp-media.h:
5320 * gst/rtsp-server/rtsp-sdp.c:
5321 * gst/rtsp-server/rtsp-stream.c:
5322 * gst/rtsp-server/rtsp-stream.h:
5323 rtsp-server: Implement clock signalling according to RFC7273
5324 For NTP and PTP clocks we signal the actual clock that is used and signal
5325 the direct media clock offset.
5326 For all other clocks we at least signal that it's the local sender clock.
5327 This allows receivers to know which clock was used to generate the media and
5328 its RTP timestamps. Receivers can then implement network synchronization,
5329 either absolute or at least relative by getting the sender clock rate directly
5330 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5332 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5334 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5336 * gst/rtsp-sink/gstrtspclientsink.c:
5337 rtspclientsink: Add support for setting the multicast interface
5338 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5340 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5342 * gst/rtsp-server/rtsp-media-factory.c:
5343 * gst/rtsp-server/rtsp-media-factory.h:
5344 * gst/rtsp-server/rtsp-media.c:
5345 * gst/rtsp-server/rtsp-media.h:
5346 * gst/rtsp-server/rtsp-stream.c:
5347 * gst/rtsp-server/rtsp-stream.h:
5348 rtsp-media: Add support for setting the multicast interface
5349 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5351 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5353 * gst/rtsp-sink/gstrtspclientsink.c:
5354 rtspclientsink: use new gst_element_class_add_static_pad_template()
5355 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5357 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5362 === release 1.8.0 ===
5364 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5370 * gst-rtsp-server.doap:
5373 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5375 * gst/rtsp-server/rtsp-stream.c:
5376 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5377 This would get us NO_PREROLL in the bin again and break seeking.
5378 Thanks to Carlos Rafael Giani for helping to debug this!
5379 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5381 === release 1.7.91 ===
5383 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5389 * gst-rtsp-server.doap:
5392 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5394 * gst/rtsp-server/rtsp-stream.c:
5395 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5396 Without this, RECORD pipelines are broken because
5397 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5398 added later. Previously it was there earlier and due to NO_PREROLL caused the
5399 pipeline to preroll immediately
5400 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5401 as the corresponding code previously was only for PLAY pipelines.
5402 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5404 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5406 * gst/rtsp-server/rtsp-stream.c:
5407 rtsp-stream: Fix typo in the docstring
5408 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5410 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5412 * gst/rtsp-server/rtsp-stream.c:
5413 rtsp-stream: Disable multicast loopback for all our sockets
5414 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5415 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5416 loopback setting on the socket... while udpsink does which unfortunately has
5417 no effect here on Windows but on Linux.
5418 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5420 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5422 * tests/check/gst/stream.c:
5423 stream tests: added new tests
5424 Test a case when the address pool only contains multicast addresses
5425 and the client is requesting unicast udp.
5426 Added tests for multicast ports allocation.
5427 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5429 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5431 * gst/rtsp-server/rtsp-stream.c:
5432 rtsp-stream: Only bind multicast sockets to ANY on Windows
5433 On Linux it is still needed to bind to the multicast address
5434 to filter out random other packets, while on Windows binding
5435 to multicast addresses just fails.
5437 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5439 * gst/rtsp-server/rtsp-stream.c:
5440 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5441 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5442 addresses, which is something that used to work before. For unicast addresses
5443 if the pool contains none, we just allocate them as if there is no pool at
5445 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5447 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5449 * gst/rtsp-server/rtsp-client.c:
5450 * gst/rtsp-server/rtsp-stream.c:
5451 rtsp-server: Fix indentation
5453 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5455 * gst/rtsp-server/rtsp-stream.c:
5456 rtsp-stream: Don't bind the sockets to multicast addresses
5457 This works on Linux but fails completely on Windows. You're supposed
5458 to bind to ANY and then join the multicast group.
5459 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5461 === release 1.7.90 ===
5463 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5469 * gst-rtsp-server.doap:
5472 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5475 Automatic update of common submodule
5476 From b64f03f to 6f2d209
5478 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5480 * gst/rtsp-sink/gstrtspclientsink.c:
5481 * tests/check/gst/rtspclientsink.c:
5482 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5483 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5485 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5487 * tests/check/gst/media.c:
5488 * tests/check/gst/rtspclientsink.c:
5489 * tests/check/gst/rtspserver.c:
5490 * tests/check/gst/stream.c:
5491 tests: unit test fixes
5492 Removed port allocation test from the media suite.
5493 The port allocation failure is now in the stream suite.
5495 Make sure that the media is suspended after the DESCRIBE request
5496 before reconfiguring the UDP sinks.
5498 In the RECORD case we have to set async property to false
5499 for the appsink element in the test in order to make sure
5500 that the media pipeline doesn't hang in start_preroll().
5501 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5503 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5505 * gst/rtsp-server/rtsp-client.c:
5506 * gst/rtsp-server/rtsp-stream.c:
5507 * gst/rtsp-server/rtsp-stream.h:
5508 rtsp-stream: postpone UDP socket allocation until SETUP
5509 Postpone the allocation of the UDP sockets until we know
5510 what transport has been chosen by the client.
5511 Both unicast and multicast UDP sources are created in one
5513 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5515 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5517 * gst/rtsp-server/rtsp-stream.c:
5518 rtsp-stream: postpone the creation of the UDP sources
5519 Code refactoring: allocate the UDP ports after the sender and
5520 the reciver parts have been created.
5521 We postpone the creation of the UDP sources until the UDP
5522 ports have been allocated.
5523 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5525 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5527 * gst/rtsp-server/rtsp-stream.c:
5528 rtsp-stream: added function for setting UDP sources to PLAYING state
5529 Code refactoring: Introduced a function for setting UDP sources
5531 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5533 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5535 * gst/rtsp-server/rtsp-stream.c:
5536 rtsp-stream: added function for creating and configuring UDP sources
5537 Code refactoring: create and configure UDP sources in a separate function.
5538 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5540 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5542 * gst/rtsp-server/rtsp-stream.c:
5543 rtsp-stream: added function for RTP/RTCP socket configuration
5544 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5545 in a separate function.
5546 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5548 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5550 * gst/rtsp-server/rtsp-stream.c:
5551 rtsp-stream: added function for creating and configuring UDP sinks
5552 Code refactoring: create and configure UDP sinks in a separate function.
5553 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5555 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5557 * gst/rtsp-server/rtsp-stream.c:
5558 rtsp-stream: added helper function for creating the sender/receiver parts
5559 Code refactoring: introduced helper function for creating
5560 the receiver and the sender parts of the streaming pipeline.
5561 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5563 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5568 === release 1.7.2 ===
5570 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5576 * gst-rtsp-server.doap:
5579 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5581 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5582 uninstalled.pc: add support for non libtool build systems
5583 Currently the .la path is provided which requires to use libtool as
5584 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5585 It is fine as long as the application is built using libtool.
5586 So currently it is not possible to compile a GStreamer application
5587 within gst-uninstalled with CMake or other build system different
5589 This patch allows to do the following in gst-uninstalled env:
5590 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5591 gstreamer-rtsp-server-1.0)
5592 Previously it required to prepend libtool --mode=link
5593 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5595 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5597 * gst/rtsp-sink/gstrtspclientsink.c:
5598 rtspclientsink: remove check for impossible condition
5599 Goto error label checks stream to see if it needs to be unreferenced before
5600 returning, but this goto jumps happens before the stream is ever set, so it
5601 will always be NULL in this error label.
5604 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5606 * gst/rtsp-sink/gstrtspclientsink.c:
5607 rtspclientsink: clean switch statements
5608 Coverity demands for fallthrough statements to be clearly commented,
5609 to distinguish from accidental fall throughs. And it also needs all
5610 cases to finish with a break, even if the break is never going to be
5611 executed like in the case of a continue jump.
5615 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5617 * tests/check/Makefile.am:
5618 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5619 To get the CK_DEFAULT_TIMEOUT defined for all tests
5620 Also removes a 120 seconds timeout that was set as default
5621 explicitly in this module
5622 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5624 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5628 Automatic update of common submodule
5629 From 86e4663 to b64f03f
5631 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5633 * gst/rtsp-server/rtsp-media.c:
5634 rtsp-media: fix state_lock not locked again when preroll fails
5635 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5637 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5640 configure: Move plugin specific flags below all the others
5641 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5642 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5644 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5646 * gst/rtsp-sink/gstrtspclientsink.c:
5647 rtspclientsink: Simplify slightly using new -base API
5648 Use the new Mikey and SDP API in the base plugins libs
5649 to simplify some code.
5650 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5652 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5657 * gst/rtsp-sink/Makefile.am:
5658 * gst/rtsp-sink/gstrtspclientsink.c:
5659 * gst/rtsp-sink/gstrtspclientsink.h:
5660 * gst/rtsp-sink/plugin.c:
5661 * tests/check/Makefile.am:
5662 * tests/check/gst/rtspclientsink.c:
5663 rtspsink: Add rtspclientsink element
5664 Add an rtspclientsink element that accepts streams for which
5665 there is a registered payloader and sends them to
5666 an RTSP server using RECORD.
5667 Sending is synchronised to the pipeline clock. Payload-types
5668 are automatically selected. The 'new-payloader' signal is fired
5669 for custom configuration of payloaders when they are created.
5670 Can now stream a movie like this:
5672 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5673 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5675 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5676 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5677 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5679 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5681 * gst/rtsp-server/rtsp-stream.c:
5682 * gst/rtsp-server/rtsp-stream.h:
5683 rtsp-stream: Add functions for using rtsp-stream from the client
5684 Add a boolean to indicate that the rtsp-stream is running on the
5685 'client' side of an RTSP connection, for sending streams via
5686 RECORD. In that case, the roles of the client/server ports
5687 in transport setup are swapped.
5688 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5690 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5692 * gst/rtsp-server/rtsp-sdp.c:
5693 * gst/rtsp-server/rtsp-sdp.h:
5694 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5695 A new function that adds info from a GstRTSPStream into an SDP message.
5696 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5698 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5700 * gst/rtsp-server/rtsp-media.c:
5701 rtsp-media: Fix mutex beeing unlocked while they should be locked
5702 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5704 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5706 * gst/rtsp-server/rtsp-media-factory.c:
5707 rtsp-media-factory: add missing break in "clock" property setter
5710 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5712 * gst/rtsp-server/rtsp-stream.c:
5713 rtsp-stream: fixed assert during update transport
5714 When RTSP server trying update transport during multicast, it throws an
5715 assert. The assert is thrown because it is trying to get the parent of
5716 an non-existing funnel element.
5717 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5719 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5721 * gst/rtsp-server/rtsp-permissions.h:
5722 * gst/rtsp-server/rtsp-thread-pool.h:
5723 * gst/rtsp-server/rtsp-token.h:
5724 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5725 gtk-doc can handle static inline functions just fine these days,
5726 there's no need for this stuff any more.
5728 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5730 * gst/rtsp-server/rtsp-media.c:
5731 * gst/rtsp-server/rtsp-sdp.c:
5732 sdp: replace duplicated codes to call new base sdp apis
5733 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5735 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5737 * examples/test-netclock.c:
5738 test-netclock: Use the new API to configure a clock directly
5740 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5742 * gst/rtsp-server/rtsp-media-factory.c:
5743 * gst/rtsp-server/rtsp-media-factory.h:
5744 * gst/rtsp-server/rtsp-media.c:
5745 * gst/rtsp-server/rtsp-media.h:
5746 rtsp-media: Add API to directly configure a clock on the media pipelines
5748 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5750 * gst/rtsp-server/rtsp-media.c:
5751 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5753 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5755 * gst/rtsp-server/rtsp-media-factory.c:
5756 rtsp-media-factory: Add FIXME for 2.0
5758 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5760 * gst/rtsp-server/rtsp-stream.c:
5761 rtsp-stream: Fix indentation
5763 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5765 * gst/rtsp-server/rtsp-media.c:
5766 rtsp-media: Do not prepare media after media times out
5767 Deferred calls to start_prepare() can be deferred past the point until
5768 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5769 prepared to wait. Previously there was no lock and no check for this
5770 situation. This meant that a media could be prepared and unprepared
5771 simultaneously by two different threads. Now a lock is in place and a
5772 suitable check is done.
5773 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5775 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5777 * gst/rtsp-server/rtsp-client.c:
5778 * gst/rtsp-server/rtsp-media-factory.c:
5779 * gst/rtsp-server/rtsp-media-factory.h:
5780 * gst/rtsp-server/rtsp-media.c:
5781 * gst/rtsp-server/rtsp-media.h:
5782 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5783 Without TEARDOWN it might be desireable to keep the media running and continue
5784 sending data to the client, even if the RTSP connection itself is
5786 Only do this for session medias that have only UDP transports. If there's at
5787 least on TCP transport, it will stop working and cause problems when the
5788 connection is disconnected.
5789 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5791 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5796 === release 1.7.1 ===
5798 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5804 * gst-rtsp-server.doap:
5807 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5810 configure: Make -Bsymbolic check work with clang.
5811 Update the -Bsymbolic check with the version glib has. This version
5813 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5815 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5817 * gst/rtsp-server/rtsp-session-pool.c:
5818 rtsp-session-pool: Avoid dollar sign ($) in session ids
5819 Live555 in VLC strips off dollar signs and then gets very confused,
5820 we don't loose too much entropy by just skipping it.
5822 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5824 * gst/rtsp-server/rtsp-address-pool.h:
5825 * gst/rtsp-server/rtsp-auth.h:
5826 * gst/rtsp-server/rtsp-client.h:
5827 * gst/rtsp-server/rtsp-media-factory-uri.h:
5828 * gst/rtsp-server/rtsp-media-factory.h:
5829 * gst/rtsp-server/rtsp-media.h:
5830 * gst/rtsp-server/rtsp-mount-points.h:
5831 * gst/rtsp-server/rtsp-permissions.h:
5832 * gst/rtsp-server/rtsp-server.h:
5833 * gst/rtsp-server/rtsp-session-media.h:
5834 * gst/rtsp-server/rtsp-session-pool.h:
5835 * gst/rtsp-server/rtsp-session.h:
5836 * gst/rtsp-server/rtsp-stream-transport.h:
5837 * gst/rtsp-server/rtsp-stream.h:
5838 * gst/rtsp-server/rtsp-thread-pool.h:
5839 * gst/rtsp-server/rtsp-token.h:
5840 rtsp-server: Add g_autoptr() support to all types
5841 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5843 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5845 * gst/rtsp-server/rtsp-stream.c:
5846 rtsp-stream: fixed valgrind error
5847 Fixed the valgrind error in unit test. The UDP source created during
5848 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5850 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5852 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5856 Automatic update of common submodule
5857 From b319909 to 86e4663
5859 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5861 * gst/rtsp-server/rtsp-client.c:
5862 rtsp-client: suspend media during setup request
5863 SETUP request from clients needs to suspend the media to clear the
5864 prerolled buffers. Otherwise it will not affect the prerolled buffer
5865 and the prerolled buffers will be incorrect (for example block-size
5866 from setup request will not affect the prerolled buffer unless the
5867 media is suspended).
5868 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5870 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5872 * gst/rtsp-server/rtsp-stream.c:
5873 rtsp-stream: create stream pipeline based on transport
5874 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5875 only UDP is set as the transport protocol, it will not add the extra tee
5876 or queue element to the pipeline. Both these elements will be added, if
5877 it supports both TCP and UDP protocols. This improves the pipeline
5878 performance when one protocol is present.
5879 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5881 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5883 * gst/rtsp-server/rtsp-stream.c:
5884 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5885 Adding them when not needed will start some logic inside rtpbin that might be
5886 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5887 would start up a rtpjitterbuffer and behave in weird ways.
5888 We still set up the UDP sources for RTP receiving for a sender media to be
5889 able to receive any packets sent by the client for NAT traversal. They will
5890 all go to a fakesink though.
5891 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5892 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5893 receive ASYNC_DONE after a seek.
5894 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5896 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5898 * gst/rtsp-server/rtsp-stream.c:
5899 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5900 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5901 Previously we were only setting this for sender sockets, which caused looped
5902 back packets to be received on Windows if a multicast transport was used.
5904 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5906 * examples/test-record-auth.c:
5907 * examples/test-record.c:
5908 examples: Actually use the provided port in the record examples
5910 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5912 * examples/test-record-auth.c:
5913 test-record-auth: Add the option to build in TLS support
5915 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5917 * examples/test-auth.c:
5918 test-auth: Use an 'anonymous' user for unauthenticated default
5919 There's a comment on one of the resources that 'user' and 'admin'
5920 shouldn't even be able to see it, but they can if the default
5921 token is 'admin2', since that gives them access anyway.
5923 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5925 * examples/.gitignore:
5926 * examples/Makefile.am:
5927 * examples/test-record-auth.c:
5928 Add test-record-auth example
5930 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5932 * gst/rtsp-server/rtsp-client.c:
5933 * tests/check/gst/client.c:
5934 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5936 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5938 * gst/rtsp-server/rtsp-server.c:
5939 rtsp-server: Change the logic so we don't pop a NULL context
5940 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5941 will sometimes fail. This call is made before any context is pushed
5942 resulting in an attempt to pop a NULL context.
5943 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5945 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5947 * tests/check/gst/rtspserver.c:
5948 rtspserver: Add udp-mcast transport SETUP test
5949 Refactor utility functions in the test file so they can handle
5950 more than UDP and TCP as lower transport.
5951 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5953 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5955 * gst/rtsp-server/rtsp-stream.c:
5956 rtsp-stream: Always unref return value of gst_object_get_parent()
5957 Fixes a leak of a GstBin in the udp-mcast case.
5958 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5960 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5963 Automatic update of common submodule
5964 From b99800a to b319909
5966 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5969 Use new GST_ENABLE_EXTRA_CHECKS #define
5970 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5972 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5975 Automatic update of common submodule
5976 From 6babecd to b99800a
5978 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5981 Update GLib dependency to 2.40.0
5983 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5985 * examples/test-mp4.c:
5986 * gst/rtsp-server/rtsp-stream.c:
5987 stream: listen to sender ssrc signals
5988 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5990 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5993 common: update for new suppression
5994 Makes check-valgrind pass with glib 2.46
5996 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5998 * gst/rtsp-server/rtsp-media.c:
5999 rtsp-media: Take reference to media that will be prepared
6000 default_prepare() takes a transfer-none reference GstRTSPMedia object.
6001 Later on a g_idle_source_new() is created and a pointer to the media
6002 object is passed as user data. If the media is freed before the idle
6003 source is dispatched the media object pointer is invalid, but the idle
6004 source callback expects it to still be valid. To fix this a reference to
6005 the media object is taken when registering the source callback function
6006 and a corresponding release of the reference is done when the souce is
6008 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
6010 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
6012 * examples/test-launch.c:
6013 * examples/test-mp4.c:
6014 * examples/test-ogg.c:
6015 * examples/test-record.c:
6016 * examples/test-uri.c:
6017 rtsp-server: Fix memory leaks when context parse fails
6018 When g_option_context_parse fails, context and error variables are not getting free'd
6019 which results in memory leaks. Free'ing the same.
6020 And replacing g_error_free with g_clear_error, which checks if the error being passed
6021 is not NULL and sets the variable to NULL on free'ing.
6022 https://bugzilla.gnome.org/show_bug.cgi?id=753863
6024 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
6029 === release 1.6.0 ===
6031 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6037 * gst-rtsp-server.doap:
6040 === release 1.5.91 ===
6042 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
6048 * gst-rtsp-server.doap:
6051 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
6053 * docs/libs/gst-rtsp-server-sections.txt:
6054 * gst/rtsp-server/rtsp-stream.c:
6055 stream: fix docs for recently-added get/set_buffer_size API
6056 https://bugzilla.gnome.org/show_bug.cgi?id=749095
6058 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
6060 * gst/rtsp-server/rtsp-media.c:
6061 rtsp-media: Don't crash on encrypted RTX SDP
6062 In parse_keymgmt(), don't mutate the input string that's been passed
6063 as const, especially since we might need the original value again if
6064 the same key info applies to multiple streams (RTX, for example).
6065 https://bugzilla.gnome.org/show_bug.cgi?id=754753
6067 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
6069 * examples/test-mp4.c:
6070 test-mp4: Support filenames with spaces in them. Error out on too few arguments
6072 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
6074 * examples/test-record.c:
6075 test-record: Check parameter count and print out help
6076 If no launch pipeline was supplied, print out some help
6078 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
6080 * gst/rtsp-server/rtsp-media.c:
6081 * gst/rtsp-server/rtsp-stream.c:
6082 * gst/rtsp-server/rtsp-stream.h:
6083 rtsp-stream: Implement UDP buffer size setting.
6084 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
6086 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
6087 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
6089 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
6091 * gst/rtsp-server/rtsp-media.h:
6092 rtsp-media: Fix small typo causing gtk-doc to complain
6094 === release 1.5.90 ===
6096 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
6102 * gst-rtsp-server.doap:
6105 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6107 * gst/rtsp-server/rtsp-media-factory.c:
6108 media-factory: get port number through gst_rtsp_url_get_port
6109 https://bugzilla.gnome.org/show_bug.cgi?id=753473
6111 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
6113 * tests/check/gst/media.c:
6114 media-test: Removing unnecessary assertion
6115 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6117 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6119 * gst/rtsp-server/rtsp-server.c:
6120 Document that source keeps a ref on server until it's destroyed
6121 https://bugzilla.gnome.org/show_bug.cgi?id=749227
6123 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6125 * tests/check/gst/media.c:
6126 media-test: Test for multiple dynamic payload
6127 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6129 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6131 * gst/rtsp-server/rtsp-media.c:
6132 media: Only add fakesink once per pipeline
6133 The intention is to prevent going PLAYING state before pads are created.
6134 If there was mutilple dynamic payload, it would leak few fakesink and
6135 actually prevent from ever reaching playing state.
6136 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6138 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6140 * gst/rtsp-server/rtsp-media.c:
6141 Revert "rtsp-media: Only add 1 fakesink per pipeline"
6142 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
6144 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-media.c:
6147 rtsp-media: Only add 1 fakesink per pipeline
6148 There should be only one fakesink per pipeline, not per dynpay. This
6149 would lead to element naming clash.
6151 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
6153 * gst/rtsp-server/rtsp-media.c:
6154 rtsp-media: assertion error due to wrong condition check
6155 In media to caps function, reserved_keys array is being used for variable i,
6156 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
6157 changed it to variable j
6158 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6160 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
6162 * gst/rtsp-server/rtsp-media.c:
6163 rtsp-media: Strip keys from the fmtp that we use internally in our caps
6164 Skip keys from the fmtp, which we already use ourselves for the
6165 caps. Some software is adding random things like clock-rate into
6166 the fmtp, and we would otherwise here set a string-typed clock-rate
6167 in the caps... and thus fail to create valid RTP caps
6168 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6170 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6172 * gst/rtsp-server/rtsp-thread-pool.c:
6173 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
6174 https://bugzilla.gnome.org/show_bug.cgi?id=752640
6176 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
6179 Automatic update of common submodule
6180 From f74b2df to 9aed1d7
6182 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
6187 === release 1.5.2 ===
6189 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
6195 * gst-rtsp-server.doap:
6198 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
6200 * gst/rtsp-server/rtsp-client.c:
6201 * gst/rtsp-server/rtsp-client.h:
6202 * tests/check/gst/client.c:
6203 rtsp-client: allow application to decide what requirements are supported
6204 Add "check-requirements" signal and vfunc to allow application
6205 (and subclasses) to check the requirements.
6206 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
6207 https://bugzilla.gnome.org/show_bug.cgi?id=749417
6209 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6212 Automatic update of common submodule
6213 From 6015d26 to f74b2df
6215 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6217 * gst/rtsp-server/rtsp-media.c:
6218 rtsp-media: Always use real payloader when creating streams
6219 A bin that contains the real payloader might be used as payloader. In this
6220 case we have to get the real payloader for the various properties it provides.
6221 Example use cases for this are bins that payload some media and then have
6222 additional elements that add metadata or RTP extension headers to the stream.
6223 https://bugzilla.gnome.org/show_bug.cgi?id=750800
6225 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
6227 * examples/test-netclock-client.c:
6228 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
6230 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
6232 * examples/test-netclock-client.c:
6233 * examples/test-netclock.c:
6234 test-netclock: Use new ntp-time-source property on rtpbin
6235 Select the clock time to be used as NTP time source. This allows proper
6236 synchronization between receivers, independent of sharing base times, and just
6237 requires them to use the same clock.
6239 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6241 * examples/test-netclock-client.c:
6242 * examples/test-netclock.c:
6243 test-netclock: Setting the same base time on sender and receiver is not necessary
6244 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
6246 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6248 * gst/rtsp-server/rtsp-stream.c:
6249 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
6250 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6252 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6254 * docs/libs/gst-rtsp-server.types:
6255 docs: add missing types
6256 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6258 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6260 * docs/libs/gst-rtsp-server-sections.txt:
6261 docs: add missing apis
6262 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6264 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
6266 * examples/test-netclock-client.c:
6267 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
6269 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6271 * docs/libs/gst-rtsp-server-sections.txt:
6272 * gst/rtsp-server/rtsp-auth.c:
6273 * gst/rtsp-server/rtsp-auth.h:
6274 GstRTSPAuth: Add client certificate authentication support
6275 https://bugzilla.gnome.org/show_bug.cgi?id=750471
6277 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
6279 * examples/test-netclock-client.c:
6280 test-netclock-client: Use new GstClock API to wait for clock synchronization
6282 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
6284 * examples/test-netclock-client.c:
6285 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
6286 A mainloop is needed to get glimagesink to display something on OSX, and
6287 the source-setup signal just makes things a little bit easier.
6289 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
6292 Automatic update of common submodule
6293 From d9a3353 to 6015d26
6295 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
6298 Automatic update of common submodule
6299 From d37af32 to d9a3353
6301 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6304 Automatic update of common submodule
6305 From 21ba2e5 to d37af32
6307 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6310 Automatic update of common submodule
6311 From c408583 to 21ba2e5
6313 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6315 * docs/libs/Makefile.am:
6316 docs: remove variables that we define in the snippet from common
6317 This is syncing our Makefile.am with upstream gtkdoc.
6319 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6322 Automatic update of common submodule
6323 From 44a3517 to c408583
6325 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6330 === release 1.5.1 ===
6332 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6338 * gst-rtsp-server.doap:
6341 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6343 * gst/rtsp-server/rtsp-client.c:
6344 rtsp-client: No flush during Teardown.
6345 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6346 backlog is empty it can happen that just a part of a message will be
6347 sent and rest is in backlog queue. If then flush during teardown
6348 just a part of message will be sent.This can lead to client miss
6349 teardown response since it expect to get the last part of message.
6350 The flushing during teardown was introduced to fix a deadlock that now
6351 is fixed more generally in handle_request by temporary setting backlog
6353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6355 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6357 * tests/check/Makefile.am:
6358 tests: Use AM_TESTS_ENVIRONMENT
6359 Needed by the new automake test runner and the
6360 current version of the common submodule.
6362 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6364 * gst/rtsp-server/rtsp-media.h:
6365 * gst/rtsp-server/rtsp-stream.h:
6366 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6368 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6370 * gst/rtsp-server/rtsp-media.c:
6371 rtsp-media: Mark some more functions static
6373 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6375 * gst/rtsp-server/rtsp-media.c:
6376 rtsp-media: Only unblock the media in suspend() when actually changing the state
6377 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6379 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6381 * examples/test-video-rtx.c:
6382 examples: Use AVPF profile for the RTX example
6384 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6386 * gst/rtsp-server/rtsp-sdp.c:
6387 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6389 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6391 * gst/rtsp-server/rtsp-stream.c:
6392 rtsp-stream: get valid clock-rate from last-sample
6393 clock-rate in last-sample's caps is integer, not unsigned.
6394 To get this value properly, variable needs to be type-casted to int.
6395 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6397 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6401 autogen.sh: only run autopoint if gettext requested in configure.ac
6402 Not just because there happens to be a po directory.
6403 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6405 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6408 Revert "configure.ac: uncomment gettext version setup"
6409 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6410 We don't need a gettext setup here and there's no po
6411 directory either, so no reason why autopoint would be
6412 run in the first place.
6413 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6415 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6417 * examples/test-multicast.c:
6418 * examples/test-multicast2.c:
6419 * examples/test-sdp.c:
6420 * examples/test-video-rtx.c:
6421 * examples/test-video.c:
6422 * tests/test-cleanup.c:
6423 * tests/test-reuse.c:
6424 Fix timeout function signatures across tests and examples
6426 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6428 * tests/check/Makefile.am:
6429 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6430 Make sure the test environment is set up.
6431 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6433 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6436 configure: bump automake requirement to 1.14 and autoconf to 2.69
6437 This is only required for builds from git, people can still
6438 build tarballs if they only have older autotools.
6439 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6441 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6444 configure.ac: uncomment gettext version setup
6445 Fixes autogen.sh. It would run autopoint, which would complain
6446 that it could not find the gettext version in configure.ac.
6447 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6449 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6451 * examples/test-video-rtx.c:
6452 test-video-rtx: set exact payload type to PCMA payloader
6453 Setting wrong payload type causes failure to do retransmission through audio stream
6454 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6456 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6458 * gst/rtsp-server/rtsp-media.c:
6459 * gst/rtsp-server/rtsp-stream.c:
6460 * gst/rtsp-server/rtsp-stream.h:
6461 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6462 Because of duplicated g_signal_connect for request-aux-sender signal,
6463 wrong stream pointer is passed to the signal handler.
6464 Instead of passing each stream, pass stream array and get the relevant stream.
6465 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6467 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6471 Update autogen.sh to latest version from common
6472 Fixes build after aclocal_check etc. helpers have been removed.
6474 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6477 Automatic update of common submodule
6478 From bc76a8b to c8fb372
6480 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6482 * gst/rtsp-server/rtsp-stream.c:
6483 rtsp-stream: Limit the queues to 1 buffer
6484 We only need them to be able to pre-roll, queueing up more data here
6485 is only going to harm latency and memory usage.
6487 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6489 * gst/rtsp-server/rtsp-stream.c:
6490 rtsp-stream: Update comment and ASCII art to the latest code
6491 We have a queue in front of the udpsink too to prevent the pipeline from
6494 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6496 * gst/rtsp-server/rtsp-stream.c:
6497 rtsp-media: Properly return first rtptime
6498 Instead we where returning first GstBuffer timestamp. This would result
6499 in clock skew and unwanted behaviour in RTSP playback.
6500 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6502 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6504 * gst/rtsp-server/rtsp-stream.c:
6505 rtsp-stream: Don't leave buffer mapped
6506 If the seq is NULL, the RTP buffer was left mapped. We should always
6509 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6514 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6516 * gst/rtsp-server/rtsp-media-factory.c:
6517 * tests/check/gst/client.c:
6518 Fix double semicolons
6520 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6522 * gst/rtsp-server/rtsp-stream.c:
6523 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6524 This gives more accurate values than asking the payloader. There might be
6525 queueing happening between the payloader and the sink.
6526 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6528 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6530 * gst/rtsp-server/rtsp-media.c:
6531 rtsp-media: Don't seek for PLAY if the position will not change
6532 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6534 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6536 * gst/rtsp-server/rtsp-media.c:
6537 rtsp-media: Don't include payload type in the caps for framesize
6538 When the sdp media attribute framesize are converted to caps
6539 the <payload> should not be included.
6540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6541 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6543 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6545 * gst/rtsp-server/rtsp-sdp.c:
6546 rtsp-sdp: add payload type to the sdp framesize attribute
6547 The sdp framesize attribute is desribed in RFC6064. It is specified
6548 for payloading of H263 and has the following form
6549 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6550 should be added to the caps in a payloader and the <payload type> should
6551 be added by the rtsp-server.
6552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6554 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6556 * examples/test-uri.c:
6557 examples: test-uri: fix tainted variable
6558 Insignificant but this keeps Coverity happy.
6561 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6563 * examples/.gitignore:
6564 * examples/Makefile.am:
6565 * examples/test-netclock-client.c:
6566 * examples/test-netclock.c:
6567 examples: Add a simple example of network synch for live streams.
6568 An example server and client that works for synchronising live streams
6569 only - as it can't support pause/play.
6571 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6573 * gst/rtsp-server/rtsp-media-factory.c:
6574 * gst/rtsp-server/rtsp-media-factory.h:
6575 rtsp-media-factory: Add functions to set/get the media gtype
6576 Allow specifying the GType of a GstRtspMedia subclass to create
6577 as a simpler way to get the factory to create a custom
6578 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6580 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6582 * gst/rtsp-server/rtsp-media.c:
6583 rtsp-media: fix double unlock in _get_buffer_size()
6584 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6585 because of double g_mutex_unlock () usage.
6586 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6588 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6590 * gst/rtsp-server/rtsp-session-pool.c:
6591 * gst/rtsp-server/rtsp-session.c:
6592 * gst/rtsp-server/rtsp-session.h:
6593 rtsp-session: Use monotonic time for RTSP session timeout
6594 Changed RTSP session timeout handling to monotonic time
6595 and deprecating the API for current system time.
6596 This fixes timeouts when the system time changes.
6597 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6599 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6601 * gst/rtsp-server/rtsp-client.c:
6602 * gst/rtsp-server/rtsp-media.c:
6603 rtsp-client: Only error out in PLAY if seeking actually failed
6604 If the media was just not seekable, we continue from whatever position we are
6605 and let the client decide if that is what is wanted or not.
6606 Only if the actual seek failed, we can't really recover and should error out.
6608 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6610 * gst/rtsp-server/rtsp-stream.c:
6611 rtsp-stream: Add necessary queues between tee and multiudpsink
6612 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6614 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6616 * gst/rtsp-server/rtsp-client.c:
6617 * gst/rtsp-server/rtsp-media.c:
6618 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6619 Instead error out properly the same way as if the SEEKING query already
6622 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6624 * gst/rtsp-server/rtsp-stream.h:
6625 rtsp-stream: minor code formatting fix
6627 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6629 * gst/rtsp-server/rtsp-media.c:
6630 rtsp-media: fix logic for collect_streams
6631 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6632 all streams it knows if it got any, and can check if the transport mode is OK.
6635 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6637 * gst/rtsp-server/rtsp-media.c:
6638 rtsp-media: Don't set the transport mode based on what elements we find
6639 Just print a warning if the one that was set before disagrees with what
6640 elements we found. It must already be set to something before as this
6641 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6642 and we would reject ANNOUNCE if the RECORD flag was not set.
6644 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6646 * tests/check/gst/rtspserver.c:
6647 tests: rtspserver: rename shadowed variable
6648 We have two different 'sink' variables here,
6649 rename one of them for clarity.
6651 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6653 * gst/rtsp-server/rtsp-client.c:
6654 rtsp-client: fix awkward if clause
6656 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6658 * examples/test-uri.c:
6659 examples: test-uri: improve uri argument handling and accept file names
6660 Print an error if the argument passed is not a URI and can't
6661 be converted into one, or no arguments have been provided.
6663 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6665 * examples/test-uri.c:
6666 examples: test-uri: don't remove mount point after 10 seconds
6667 It's very irritating when trying to test stuff repeatedly
6668 and serves no real purpose other than showing that it can
6671 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6673 * examples/.gitignore:
6674 examples: add new test-record to .gitignore
6676 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6678 * examples/test-record.c:
6679 * gst/rtsp-server/rtsp-client.c:
6680 * gst/rtsp-server/rtsp-media-factory.c:
6681 * gst/rtsp-server/rtsp-media-factory.h:
6682 * gst/rtsp-server/rtsp-media.c:
6683 * gst/rtsp-server/rtsp-media.h:
6684 * tests/check/gst/rtspserver.c:
6685 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6687 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6689 * examples/test-record.c:
6690 test-record: Set latency for playback-style example to 2s instead of 200ms
6692 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6694 * tests/check/gst/rtspserver.c:
6695 tests: add some unit tests for ANNOUNCE and RECORD
6696 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6698 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6700 * gst/rtsp-server/rtsp-client.c:
6701 rtsp-client: fix a couple of leaks in handle_announce
6703 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6705 * gst/rtsp-server/rtsp-media-factory.c:
6706 * gst/rtsp-server/rtsp-media-factory.h:
6707 * gst/rtsp-server/rtsp-media.c:
6708 * gst/rtsp-server/rtsp-media.h:
6709 rtsp-media: Expose latency setting for setting the rtpbin latency
6711 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6713 * examples/test-record.c:
6714 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6716 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6718 * gst/rtsp-server/rtsp-stream.c:
6719 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6721 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6723 * examples/Makefile.am:
6724 * examples/test-record.c:
6725 * gst/rtsp-server/rtsp-client.c:
6726 * gst/rtsp-server/rtsp-client.h:
6727 * gst/rtsp-server/rtsp-media-factory.c:
6728 * gst/rtsp-server/rtsp-media-factory.h:
6729 * gst/rtsp-server/rtsp-media.c:
6730 * gst/rtsp-server/rtsp-media.h:
6731 * gst/rtsp-server/rtsp-session-media.c:
6732 * gst/rtsp-server/rtsp-stream.c:
6733 * gst/rtsp-server/rtsp-stream.h:
6734 Add initial support for RECORD
6735 We currently only support media that is RECORD or PLAY only, not both at once.
6736 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6738 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6740 * gst/rtsp-server/rtsp-stream.c:
6741 rtsp-stream: RTCP and RTP transport cache cookies seperated
6742 RTCP packets were not sent because the same tr_cache_cookie was used for
6743 both RTP and RTCP. So only one of the tr_cache lists were populated
6744 depending on which one was sent first. If the tr_cache list is not
6745 populated then no packets can be sent. Most often this happened to be
6746 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6747 resulted in both the tr_cache_lists to be populated regardless of which
6749 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6751 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6753 * gst/rtsp-server/rtsp-stream.c:
6754 rtsp-stream: fix false compiler warning
6755 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6757 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6759 * gst/rtsp-server/rtsp-client.c:
6760 rtsp-client: log interleaved data received
6762 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6764 * gst/rtsp-server/rtsp-client.c:
6765 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6767 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6769 * gst/rtsp-server/rtsp-client.c:
6770 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6772 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6774 * gst/rtsp-server/rtsp-client.c:
6775 rtsp-client: Use a random session ID in the SDP
6776 RFC4566 Section 5.2 says that it should make the username, session id,
6777 nettype, addrtype and unicast address tuple globally unique. Always using
6778 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6779 Instead let's create a 64 bit random number, which at least brings us
6780 closer to the goal of global uniqueness.
6781 https://tools.ietf.org/html/rfc4566#section-5.2
6783 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6785 * examples/test-launch.c:
6786 * examples/test-mp4.c:
6787 * examples/test-ogg.c:
6788 * examples/test-uri.c:
6789 examples: Don't call gst_init() and gst_get_option_group()
6790 The latter calls the former at the appropriate time.
6792 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6794 * gst/rtsp-server/rtsp-client.c:
6795 rtsp-client: Drop trailing \0 of RTSP DATA messages
6796 We add a trailing \0 in GstRTSPConnection to make parsing of
6797 string message bodies easier (e.g. the SDP from DESCRIBE) but
6798 for actual data this means we have to drop it or otherwise
6799 create invalid data.
6801 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6803 * gst/rtsp-server/rtsp-stream.c:
6804 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6805 Fixes crash when two threads access handle_new_sample() at the same
6806 time, one for RTP, one for RTCP.
6807 Otherwise, when iterating over the transports cache, it might be modified by
6808 another thread at the same time if the transports cookie has changed.
6809 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6811 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6813 * gst/rtsp-server/rtsp-stream.c:
6814 rtsp-stream: Set format=TIME on our app sources for TCP
6816 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6818 * gst/rtsp-server/rtsp-session-pool.c:
6819 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6820 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6821 RFC 2326 states that session IDs may consist of alphanumeric as well as
6822 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6823 Previously the session ID was URI-escaped, this meant that any character
6824 which was not alphanumeric or any of the characters +-._~ would be
6825 percent encoded. While the RFC (surprisingly) mentions that linear white
6826 space in session IDs should be URI-escaped, it does not say anything
6827 about other characters. Moreover no white space is allowed in the
6828 session ID. Finally the percent character which is the result of
6829 URI-escaping is not allowed in a session ID.
6830 So there is no reason to do any URI-escaping, and now it is removed.
6831 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6833 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6836 Automatic update of common submodule
6837 From f2c6b95 to bc76a8b
6839 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6842 Fix 'make check' from top-level directory
6844 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6846 * examples/test-launch.c:
6847 * examples/test-mp4.c:
6848 * examples/test-ogg.c:
6849 * examples/test-uri.c:
6850 examples: Add command-line parsing and take a 'port' argument
6851 This allows users to run multiple servers on different ports for testing.
6852 Only done for examples that actually take arguments and hence are capable of
6853 outputting different streams for each instance on each port.
6854 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6856 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6858 * gst/rtsp-server/rtsp-client.c:
6859 * gst/rtsp-server/rtsp-client.h:
6860 rtsp-client: Add a send_message default signal handler
6861 This allows subclasses to easily hook into the response sending
6862 mechanism without doing everything from a signal, which seems
6863 awkward from subclasses.
6865 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6868 Automatic update of common submodule
6869 From ef1ffdc to f2c6b95
6871 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6875 configure: add --disable-examples switch
6876 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6878 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6880 * examples/.gitignore:
6881 * examples/Makefile.am:
6882 * examples/test-video-rtx.c:
6883 examples: add a retransmisison example implementing RFC4588
6884 Currently only SSRC-multiplexed rtx streams are supported
6886 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6888 * gst/rtsp-server/rtsp-stream.c:
6889 rtsp-stream: Fix some minor memory leaks
6891 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6893 * gst/rtsp-server/rtsp-media.c:
6894 rtsp-media: Some minor cleanup
6896 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6898 * gst/rtsp-server/rtsp-stream.c:
6899 rtsp-stream: Fix compiler warnings
6900 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6901 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6903 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6904 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6907 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6909 * docs/libs/gst-rtsp-server-sections.txt:
6910 * gst/rtsp-server/rtsp-media-factory.c:
6911 * gst/rtsp-server/rtsp-media-factory.h:
6912 * gst/rtsp-server/rtsp-media.c:
6913 * gst/rtsp-server/rtsp-media.h:
6914 * gst/rtsp-server/rtsp-sdp.c:
6915 * gst/rtsp-server/rtsp-stream.c:
6916 * gst/rtsp-server/rtsp-stream.h:
6917 media: implement ssrc-multiplexed retransmission support
6918 based off RFC 4588 and the server-rtpaux example in -good
6920 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-stream-transport.c:
6924 * gst/rtsp-server/rtsp-stream.c:
6925 rtsp: Ref transports in hash table.
6926 Also ref streams for transports.
6927 This solves a crash when reciving a rtcp after teardown but before
6929 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6931 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6934 Automatic update of common submodule
6935 From 7bb2bce to ef1ffdc
6937 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6939 * gst/rtsp-server/rtsp-client.c:
6940 client: refactor cleanup of cached media
6942 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6944 * tests/check/gst/client.c:
6946 The session leak is now fixed, lets remove those FIXME comments.
6948 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6950 * tests/check/gst/rtspserver.c:
6951 tests: Test to setup two sessions on one connection
6952 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6954 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6956 * tests/check/gst/rtspserver.c:
6957 tests: Test setup with tcp transport
6958 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6960 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6962 * gst/rtsp-server/rtsp-client.c:
6963 client: Configure transport after creating session media
6964 The default implementation of configure_client_transport() in
6965 rtsp-client uses the session media when it chooses channels for
6966 interleaved traffic.
6967 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6969 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6971 * gst/rtsp-server/rtsp-client.c:
6972 * gst/rtsp-server/rtsp-session-media.c:
6973 client: Stop caching media in client when doing setup
6974 If the media has been managed by a session media, it should not be
6975 cached in the client any longer. The GstRTSPSessionMedia object is now
6976 responsible for unpreparing the GstRTSPMedia object using
6977 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6979 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6981 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6983 * gst/rtsp-server/rtsp-stream.c:
6984 rtsp-stream: unref srtp decoder when leaving bin
6985 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6987 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6989 * gst/rtsp-server/rtsp-client.c:
6990 rtsp-client: mikey memory leaks
6991 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6993 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6996 Automatic update of common submodule
6997 From 84d06cd to 7bb2bce
6999 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
7002 Parallelise 'make check-valgrind'
7004 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
7007 Automatic update of common submodule
7008 From a8c8939 to 84d06cd
7010 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
7013 Automatic update of common submodule
7014 From 36388a1 to a8c8939
7016 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7018 * gst/rtsp-server/rtsp-media.c:
7019 rtsp-media: deactivate media when shutting down from paused
7020 This was only done when going directly from playing.
7021 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
7023 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7025 * gst/rtsp-server/rtsp-client.c:
7026 * gst/rtsp-server/rtsp-context.h:
7027 rtsp-client: add stream transport to context
7028 We add the stream transport to the context so we can get the configured
7029 client stream transport in the setup request signal.
7030 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
7032 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7034 * gst/rtsp-server/rtsp-stream.c:
7035 stream: release lock even not all transports have been removed
7036 We don't want to keep the lock even we return FALSE because not all the
7037 transports have been removed. This could lead into a deadlock.
7038 https://bugzilla.gnome.org/show_bug.cgi?id=737797
7040 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
7042 * gst/rtsp-server/rtsp-sdp.c:
7043 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
7044 These were renamed in GstRTPBasePayload in 1.0
7046 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7048 * gst/rtsp-server/rtsp-client.c:
7049 client: set session media to NULL without the lock
7050 We need to set session medias to NULL without the client lock otherwise
7051 we can end up in a deadlock if another thread is waiting for the lock
7052 and media unprepare is also waiting for that thread to end.
7053 https://bugzilla.gnome.org/show_bug.cgi?id=737690
7055 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
7057 * gst/rtsp-server/rtsp-media.c:
7058 rtsp-media: Set state to UNPREPARING in all cases
7060 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
7062 * gst/rtsp-server/rtsp-media.c:
7063 media: set state to unpreparing when unprepare is initiated
7064 https://bugzilla.gnome.org/show_bug.cgi?id=737675
7066 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
7068 * gst/rtsp-server/rtsp-client.c:
7069 rtsp-client: Remove backlog limit while processings requests
7070 If the backlog limit is kept two cases of deadlocks may be
7071 encountered when streaming over TCP. Without the backlog
7072 limit this deadlocks can not happen, at the expence of
7074 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
7076 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
7078 * gst/rtsp-server/rtsp-client.c:
7079 rtsp-client: do not free main context before rtsp watch
7080 https://bugzilla.gnome.org/show_bug.cgi?id=737110
7082 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
7084 * tests/check/gst/rtspserver.c:
7085 tests: Extend unit test timeout to accomodate for valgrind
7086 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7088 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
7090 * gst/rtsp-server/rtsp-client.c:
7091 * gst/rtsp-server/rtsp-session.c:
7092 * gst/rtsp-server/rtsp-stream-transport.c:
7093 rtsp-*: Treat sending packets to clients as keepalive
7094 As long as gst-rtsp-server can successfully send RTP/RTCP data to
7095 clients then the client must be reading. This change makes the server
7096 timeout the connection if the client stops reading.
7097 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7099 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
7101 * gst/rtsp-server/rtsp-client.c:
7102 rtsp-client: Allow backlog to grow while expiring session
7103 Allow the send backlog in the RTSP watch to grow to unlimited size while
7104 attempting to bring the media pipeline to NULL due to a session
7105 expiring. Without this change the appsink element cannot change state
7106 because it is blocked while rendering data in the new_sample callback.
7107 This callback will block until it has successfully put the data into the
7108 send backlog. There is a chance that the send backlog is full at this
7109 point which means that the callback may block for a long time, possibly
7110 forever. Therefore the media pipeline may also be prevented from
7111 changing state for a long time.
7112 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7114 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
7116 * gst/rtsp-server/rtsp-client.c:
7117 rtsp-client: Make old compilers happy
7118 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
7119 Just in case that guint8 doesn't fit in a pointer. Just in case ...
7121 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
7123 * gst/rtsp-server/rtsp-client.c:
7124 client: raise the backlog limits before pausing
7125 We need to raise the backlog limits before pausing the pipeline or else
7126 the appsink might be blocking in the render method in wait_backlog() and
7127 we would deadlock waiting for paused.
7128 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
7130 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
7132 * gst/rtsp-server/rtsp-client.c:
7133 client: make define for the WATCH_BACKLOG
7134 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
7136 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
7138 * gst/rtsp-server/rtsp-client.c:
7139 client: simplify session transport handling
7140 link/unlink of the transport in a session was done to keep track of all
7141 TCP transports and to send RTP/RTCP data to the streams. We can simplify
7142 that by putting all the TCP transports in a hashtable indexed with the
7144 We also don't need to link/unlink the transports when we pause/resume
7145 the streams. The same effect is already achieved when we pause/play the
7146 media. Indeed, when we pause the media, the transport is removed from
7147 the media and the callbacks will not be called anymore.
7148 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
7150 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
7152 * gst/rtsp-server/rtsp-stream-transport.c:
7153 * gst/rtsp-server/rtsp-stream-transport.h:
7154 stream-transport: make method to handle received data
7155 Make a method to handle the data received on a channel. It sends the
7156 data to the stream of the transport on the RTP or RTCP pads based on
7159 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
7161 * examples/test-mp4.c:
7162 test: add example of dumping RTCP reports
7164 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
7166 * gst/rtsp-server/rtsp-media.c:
7167 * gst/rtsp-server/rtsp-stream.c:
7168 * gst/rtsp-server/rtsp-stream.h:
7169 rtsp-media: Make sure that sequence numbers are monotonic after pause
7170 The sequence number is not monotonic for RTP packets after pause. The
7171 reason is basepayloader generates a randon sequence number when the
7172 pipeline goes from ready to pause. With this fix generation of sequence
7173 number will be monotonic when going from pause to play request.
7174 https://bugzilla.gnome.org/show_bug.cgi?id=736017
7176 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
7178 * gst/rtsp-server/rtsp-client.c:
7179 rtsp-client: Protect saved clients watch with a mutex
7180 Fixes a crash when close() is called while merging clients
7181 in handle_tunnel(). In that case close() would destroy the
7182 watch while it is still being used in handle_tunnel().
7183 https://bugzilla.gnome.org/show_bug.cgi?id=735570
7185 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
7187 * gst/rtsp-server/rtsp-stream.c:
7188 rtsp-stream: Remove the multicast group udp sources when removing from the bin
7190 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
7192 * gst/rtsp-server/rtsp-media.c:
7193 * gst/rtsp-server/rtsp-stream.c:
7194 * gst/rtsp-server/rtsp-stream.h:
7195 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
7196 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
7197 seeking and will always continue counting the time. This leads to
7198 the NPT after a backwards seek to be something completely different
7199 to the actual seek position.
7200 https://bugzilla.gnome.org/show_bug.cgi?id=732644
7202 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
7204 * examples/test-appsrc.c:
7205 examples: fix another reference leak
7206 gst_rtsp_media_get_element() returns a new ref.
7208 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
7210 * examples/test-appsrc.c:
7211 examples: unref element after usage
7212 gst_bin_get_by_name_recurse_up() returns an element
7213 reference that must be unreffed after usage.
7214 https://bugzilla.gnome.org/show_bug.cgi?id=734546
7216 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
7218 * gst/rtsp-server/rtsp-media.c:
7219 signals: Fix copy-pasto in target-state signal offset
7221 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
7225 Makefile: Add usage of build-checks step
7226 Allows building checks without running them
7228 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
7230 * gst/rtsp-server/rtsp-stream.c:
7231 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
7232 When a UDP multicast transport is used it is expected that the server listens
7233 for RTP and RTCP packets on the multicast group with the corresponding port.
7234 Without this we will never get RTCP packets from clients in multicast mode.
7235 https://bugzilla.gnome.org/show_bug.cgi?id=732238
7237 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
7242 === release 1.4.0 ===
7244 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7250 * gst-rtsp-server.doap:
7253 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
7255 * gst/rtsp-server/rtsp-media.h:
7256 media: correct misspelled words in description
7257 https://bugzilla.gnome.org/show_bug.cgi?id=733244
7259 === release 1.3.91 ===
7261 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
7267 * gst-rtsp-server.doap:
7270 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
7272 * docs/libs/gst-rtsp-server-sections.txt:
7275 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
7277 * gst/rtsp-server/rtsp-server.c:
7278 server: implement client REMOVE filter
7280 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
7282 * gst/rtsp-server/rtsp-client.c:
7283 * gst/rtsp-server/rtsp-client.h:
7284 client: expose _close() method
7285 Expose a previously internal close method to close the client
7288 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
7290 * gst/rtsp-server/rtsp-session-pool.c:
7291 session-pool: signal session-removed outside of the lock
7292 Release the lock before emiting the session-removed signal.
7294 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
7296 * gst/rtsp-server/rtsp-client.c:
7297 * gst/rtsp-server/rtsp-server.c:
7298 * gst/rtsp-server/rtsp-session-pool.c:
7299 * gst/rtsp-server/rtsp-session.c:
7300 * gst/rtsp-server/rtsp-stream.c:
7301 filter: Release lock in filter functions
7302 Release the object lock before calling the filter functions. We need to
7303 keep a cookie to detect when the list changed during the filter
7304 callback. We also keep a hashtable to make sure we only call the filter
7305 function once for each object in case of concurrent modification.
7306 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7308 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7310 * gst/rtsp-server/rtsp-client.c:
7311 client: check if watch is set in handle_teardown()
7312 The unit tests run without a watch
7314 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7316 * tests/check/gst/client.c:
7317 client tests: send teardown to cleanup session
7319 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7321 * tests/check/gst/rtspserver.c:
7322 server tests: send teardown to cleanup session
7324 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7326 * gst/rtsp-server/rtsp-client.c:
7327 client: keep ref to client for the session removed handler
7328 This extra ref will be dropped when all client sessions have been
7329 removed. A session is removed when a client sends teardown, closes its
7330 endpoint of the TCP connection or the sessions expires.
7331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7333 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7335 * gst/rtsp-server/rtsp-client.c:
7336 * gst/rtsp-server/rtsp-session.c:
7337 * tests/check/gst/client.c:
7338 client: manage media in session as a last step
7339 Once we manage a media in a session, we can't unmanage it anymore
7340 without destroying it. Therefore, first check everything before we
7341 manage the media, otherwise if something is wrong we have no way to
7343 If we created a new session and something went wrong, remove the session
7344 again. Fixes a leak in the unit test.
7346 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7348 * examples/test-mp4.c:
7349 * examples/test-ogg.c:
7350 examples: print 'stream ready at url' for mp4 and ogg example
7352 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7354 * gst/rtsp-server/rtsp-client.c:
7355 * gst/rtsp-server/rtsp-sdp.c:
7356 rtsp: fix for MIKEY api change
7358 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7360 * gst/rtsp-server/rtsp-client.c:
7361 client: free watch context only once
7362 The watch context is freed when the source is destroyed. Avoids
7363 a CRITICAL when we try to unref the context twice.
7365 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7367 * gst/rtsp-server/rtsp-client.c:
7370 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7372 * gst/rtsp-server/rtsp-client.c:
7373 client: protect sessions with lock
7374 Protect the list of sessions with the lock.
7375 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7377 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7379 * gst/rtsp-server/rtsp-client.c:
7380 Client: keep a ref to the session
7381 Don't just keep a weak ref to the session objects but use a hard ref. We
7382 will be notified when a session is removed from the pool (expired) with
7383 the new session-removed signal.
7384 Don't automatically close the RTSP connection when all the sessions of
7385 a client are removed, a client can continue to operate and it can create
7386 a new session if it wants. If you want to remove the client from the
7387 server, you have to use gst_rtsp_server_client_filter() now.
7388 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7389 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7391 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7393 * gst/rtsp-server/rtsp-session-pool.c:
7394 * gst/rtsp-server/rtsp-session-pool.h:
7395 session-pool: add session-removed signal
7396 Add a signal to be notified when a session is removed from the pool.
7398 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7400 * gst/rtsp-server/Makefile.am:
7401 * gst/rtsp-server/rtsp-server.h:
7402 Make rtsp-server.h a single-include header, use it for G-I
7403 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7405 === release 1.3.90 ===
7407 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7413 * gst-rtsp-server.doap:
7416 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7418 * gst/rtsp-server/rtsp-stream.c:
7419 stream: crypto can be NULL
7421 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7423 * gst/rtsp-server/rtsp-client.c:
7424 * gst/rtsp-server/rtsp-media.c:
7425 * gst/rtsp-server/rtsp-mount-points.c:
7426 introspection: add missing allow-none annotations
7427 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7429 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7431 * gst/rtsp-server/rtsp-address-pool.c:
7432 * gst/rtsp-server/rtsp-media.c:
7433 * gst/rtsp-server/rtsp-session-media.c:
7434 * gst/rtsp-server/rtsp-session-pool.c:
7435 * gst/rtsp-server/rtsp-stream-transport.c:
7436 * gst/rtsp-server/rtsp-stream.c:
7437 * gst/rtsp-server/rtsp-token.c:
7438 introspection: add (nullable) annotations to return values
7439 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7441 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7443 * gst/rtsp-server/rtsp-client.c:
7444 * gst/rtsp-server/rtsp-stream.c:
7445 gi: improve annotations
7446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7448 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7450 * gst/rtsp-server/rtsp-client.c:
7451 * gst/rtsp-server/rtsp-media-factory.c:
7452 * gst/rtsp-server/rtsp-media.c:
7453 * gst/rtsp-server/rtsp-server.c:
7454 signals: use generic marshal function
7455 Use the generic C marshal function.
7456 Use more explicit type instead of G_TYPE_POINTER
7458 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7460 * gst/rtsp-server/rtsp-context.h:
7461 context: add type macro
7463 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7465 * gst/rtsp-server/rtsp-client.c:
7466 * gst/rtsp-server/rtsp-sdp.c:
7467 * gst/rtsp-server/rtsp-sdp.h:
7468 sdp: hide key length defines
7469 They don't have a namespace.
7471 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7476 === release 1.3.3 ===
7478 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7484 * gst-rtsp-server.doap:
7487 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7489 * gst/rtsp-server/rtsp-client.c:
7490 * gst/rtsp-server/rtsp-sdp.c:
7491 * gst/rtsp-server/rtsp-sdp.h:
7492 mikey: add different key length parameters
7493 Add encryption and authentication key length parameters to MIKEY. For
7494 the encoders, the key lengths are obtained from the cipher and auth
7495 algorithms set in the caps. For the decoders, they are obtained while
7496 parsing the key management from the client.
7497 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7499 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7501 * tests/check/gst/stream.c:
7502 stream tests: Make sure we get right multicast address from stream
7503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7505 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7507 * gst/rtsp-server/rtsp-client.c:
7508 client: ref the context until rtsp watch is alive
7509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7511 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7513 * gst/rtsp-server/rtsp-client.c:
7514 client: Destroy the rtsp watch after connection close
7516 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7518 * gst/rtsp-server/rtsp-media.c:
7519 media: fix confusing comment
7521 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7523 * gst/rtsp-server/rtsp-session.c:
7524 rtsp-session: Timeout in header.
7525 Adding the possbilty to always have timout in header.
7526 This is configurabe with setting "timeout-always-visible".
7527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7529 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7534 === release 1.3.2 ===
7536 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7543 * gst-rtsp-server.doap:
7546 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7549 Automatic update of common submodule
7550 From 211fa5f to 1f5d3c3
7552 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7554 * gst/rtsp-server/rtsp-client.c:
7555 client: store TCP ports in transport
7556 Store the TCP ports in the transport when we are doing RTSP over TCP.
7557 This way, we can easily get to the ports from the transport.
7558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7560 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7562 * gst/rtsp-server/rtsp-stream.c:
7563 stream: add signals for new RTP/RTCP encoders
7564 New signals to allow the user to configure the dynamically created
7566 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7568 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7570 * gst/rtsp-server/rtsp-media.c:
7571 * gst/rtsp-server/rtsp-media.h:
7572 media: Make suspend()/unsuspend() virtual
7573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7575 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7577 * gst/rtsp-server/rtsp-client.c:
7578 client: fix send-message signal marshaller
7579 Use generic marshalling for the send-message signal. It has
7580 two POINTER arguments, not just one.
7581 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7583 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7585 * tests/check/gst/media.c:
7586 tests: add and remove pads only once
7587 In this test we simulate a dynamic pad by watching the caps event.
7588 Because of renegotiation in the base payloader now, this caps is sent
7589 multiple times but we can only deal with 1 invocation, use a variable to
7590 only 'add and remove' the pad once.
7592 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7594 * tests/check/gst/rtspserver.c:
7595 tests: add unit test for correct handling of Require headers
7596 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7598 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7600 * gst/rtsp-server/rtsp-client.c:
7601 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7602 Servers must handle Require headers and must report a failure
7603 if they don't handle any of the Required options, see RFC 2326,
7604 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7605 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7607 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7612 === release 1.3.1 ===
7614 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7620 * gst-rtsp-server.doap:
7623 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7626 Automatic update of common submodule
7627 From bcb1518 to 211fa5f
7629 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7634 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7636 * tests/check/gst/sessionmedia.c:
7637 tests: fix memory leak in sessionmedia unit test
7639 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7641 * gst/rtsp-server/rtsp-client.c:
7642 client: emit a signal before sending a message
7643 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7645 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7647 * gst/rtsp-server/rtsp-client.c:
7648 client: pass context to send_message
7649 Pass the current context to send_message, we will need it later.
7651 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7653 * gst/rtsp-server/rtsp-client.c:
7654 client: fix typo in comment
7656 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7658 * gst/rtsp-server/rtsp-media.c:
7659 media: Do not stop thread twice if default_prepare() fails
7661 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7663 * gst/rtsp-server/rtsp-client.c:
7664 client: set the watch to flushing before going to NULL
7665 First set the watch to flushing so that we unblock any current and
7666 future attempt to send data on the watch, Then set the pipeline to
7668 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7670 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7672 * gst/rtsp-server/rtsp-session-pool.c:
7673 * tests/check/gst/sessionpool.c:
7674 rtsp-session-pool: Fixes annotation
7675 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7676 in the sessionpool test.
7677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7679 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7681 * gst/rtsp-server/rtsp-media.c:
7682 * gst/rtsp-server/rtsp-media.h:
7683 media: make media_prepare virtual
7684 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7686 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7688 * gst/rtsp-server/rtsp-media.c:
7689 * tests/check/gst/media.c:
7690 media: stop the thread in more error cases
7692 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7694 * gst/rtsp-server/rtsp-media.c:
7695 * tests/check/gst/media.c:
7696 media: allow NULL as the thread
7697 Use the default context whan passing a NULL thread.
7699 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7701 * gst/rtsp-server/rtsp-client.c:
7702 rtsp-client: indent cleanup
7703 Coverity was moaning about unreachable code, and I think it was just
7704 confused by { being before the label. We'll see if it pops up again.
7707 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7709 * gst/rtsp-server/rtsp-client.c:
7710 * gst/rtsp-server/rtsp-media.c:
7711 client: Add drop-backlog property
7712 When we have too many messages queued for a client (currently hardcoded
7713 to 100) we overflow and drop the messages. Add a drop-backlog property
7714 to control this behaviour. Setting this property to FALSE will retry
7715 to send the messages to the client by waiting for more room in the
7717 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7719 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7721 * gst/rtsp-server/rtsp-client.c:
7722 client: support for POST before GET when setting up a tunnel
7724 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7726 * gst/rtsp-server/rtsp-client.c:
7727 client: remove watch of the second client after http tunnel setup
7728 The second client will be freed after the HTTP tunnel has been set up.
7729 Make sure it's RTSP watch is never dispatched again.
7730 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7732 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7734 * gst/rtsp-server/rtsp-media.c:
7735 * tests/check/gst/media.c:
7736 media: Make media_prepare() fail if port allocation fails
7737 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7739 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7741 * tests/check/gst/media.c:
7742 media test: cleanup the thread pool in tests
7744 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7746 * gst/rtsp-server/rtsp-media.c:
7747 * tests/check/gst/media.c:
7748 rtsp-media: Unblock blocked streams in unprepare
7749 The streams will be blocked when a live media is prepared.
7750 The streams should be unblocked in gst_rtsp_media_unprepare.
7751 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7753 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7755 * gst/rtsp-server/rtsp-media.c:
7756 media: release the state lock when going to NULL
7757 Set our state to UNPREPARING and release the state-lock before
7758 setting the pipeline to the NULL state. This way, any pad-added
7759 callback will be able to take the state-lock and check that we are now
7760 unpreparing instead of deadlocking.
7761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7763 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7765 * gst/rtsp-server/rtsp-media.c:
7766 media: protect status with lock
7767 Make sure we only update the status with the lock.
7769 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7771 * gst/rtsp-server/rtsp-client.c:
7772 * gst/rtsp-server/rtsp-sdp.c:
7773 rtsp: update for MIKEY API changes
7775 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7777 * gst/rtsp-server/rtsp-client.c:
7778 client: parse the mikey response from the client
7779 Parse the mikey response from the client and update the policy for
7782 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7784 * gst/rtsp-server/rtsp-stream.c:
7785 * gst/rtsp-server/rtsp-stream.h:
7786 stream: add method to set crypto info
7787 Make a method to configure the crypto information of a stream.
7788 Set udpsrc in READY instead of PAUSED so that we can configure caps
7791 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7793 * gst/rtsp-server/rtsp-client.c:
7794 client: cleanup error paths
7796 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7798 * gst/rtsp-server/rtsp-media.c:
7801 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7803 * examples/test-video.c:
7804 test: enable SRTP only on RTSPS
7805 We only want to enable SRTP when doing rtsp over TLS so that we can
7806 exchange the keys in a secure way.
7808 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7810 * examples/test-video.c:
7811 test: print an error on failure
7813 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7816 * examples/test-video.c:
7817 * gst/rtsp-server/rtsp-sdp.c:
7818 * gst/rtsp-server/rtsp-stream.c:
7819 * tests/check/Makefile.am:
7820 stream: add SRTP support
7821 Install srtp encoder and decoder elements in rtpbin
7824 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7826 * tests/check/Makefile.am:
7827 * tests/check/gst/sessionpool.c:
7828 tests: Add unit tests for sessionpool
7829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7831 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7833 * tests/check/gst/threadpool.c:
7834 tests: Improve code coverage of rtsp-threadpool tests
7835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7837 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7839 * tests/check/gst/sessionmedia.c:
7840 tests: Improve code coverage for rtsp-session-media
7841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7843 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7845 gobject-introspection: Add annotations to support language bindings
7846 In addition a few cosmetic changes:
7847 * Adjust the order of arguments
7848 * Fix typo: occured -> occurred
7849 * Fix indentation after Return:-clauses
7850 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7852 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7854 * gst/rtsp-server/rtsp-stream.c:
7855 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7856 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7858 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7860 * gst/rtsp-server/rtsp-stream.c:
7861 stream: take caps after the session manager
7862 Take the caps for the SDP after they leave the rtpbin so that we can
7863 also get the properties added by rtpbin elements.
7865 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7867 * gst/rtsp-server/rtsp-stream.c:
7868 stream: release lock while pushing out packets
7869 Keep a cache of the transports and use this to iterate the transport
7870 while pushing packets. This allows us to release the lock early.
7871 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7873 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7875 * gst/rtsp-server/rtsp-client.c:
7876 * gst/rtsp-server/rtsp-client.h:
7877 rtsp-client: vmethod for modifying tunnel GET response
7878 Add a vmethod tunnel_http_response where the response to the HTTP GET
7879 for tunneled connections can be modified.
7880 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7882 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7884 * gst/rtsp-server/rtsp-sdp.c:
7885 sdp: make 1 media line per profile
7886 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7887 line in the SDP for each profile. The client is then supposed to pick
7888 one of the profiles in the SETUP request. Because the m= lines have the
7889 same pt, the client also knows that only 1 option is possible.
7891 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7893 * gst/rtsp-server/rtsp-media-factory.c:
7894 * gst/rtsp-server/rtsp-media-factory.h:
7895 * gst/rtsp-server/rtsp-media.c:
7896 factory: add profile property and pass to media and streams
7898 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7900 * examples/test-multicast.c:
7901 * gst/rtsp-server/rtsp-sdp.c:
7902 sdp: pass multicast connection for multicast-only stream
7903 Pass the multicast address of the stream in the connection info in the
7904 SDP so that clients try a multicast connection first.
7905 Only allow multicast connections in the test-multicast example. Also
7906 increase the TTL a little.
7908 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7911 .gitignore: Ignore gcov intermediate files
7912 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7914 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7916 * gst/rtsp-server/rtsp-stream.c:
7917 stream: release some locks in error cases
7919 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7921 docs: Enable and fix gtk-doc warnings
7922 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7923 * addresspool/mediafactory: Add missing annotation colon
7924 * stream: Annotate return value
7925 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7927 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7930 Automatic update of common submodule
7931 From fe1672e to bcb1518
7933 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7936 Automatic update of common submodule
7937 From 1a07da9 to fe1672e
7939 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7941 * examples/Makefile.am:
7942 examples: use LDADD for libs instead of LDFLAGS
7944 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7947 configure: make sure releases are in .doap file
7949 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7951 * examples/test-cgroups.c:
7952 examples: test-cgroups: don't put code with side effects into g_assert()
7953 The g_assert() might get compiled out with the right
7954 compiler/preprocessor flags.
7956 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7958 * examples/.gitignore:
7959 examples: add cgroup test binary to .gitignore
7961 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7963 * examples/test-cgroups.c:
7964 examples: fix cgroup test build
7965 Fixes build failure caused by compiler warning:
7966 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7968 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7971 .gitignore: ignore temp files created in the course of 'make check'
7973 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7975 * gst/rtsp-server/rtsp-media.c:
7976 rtsp-media: don't loose frames handling new PLAY request
7977 If client supplied a range check if the range specifies the start point.
7978 If not, then do an accurate seek to the current position. If a start
7979 point was specified do do a key unit seek to make sure the streaming
7980 starts with decodeable frames.
7981 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7983 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7985 * gst/rtsp-server/rtsp-media.c:
7986 Revert "media: only flush when setting a new start position"
7987 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7988 We need to do the flush in all cases, demuxer block currently for
7991 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7993 * gst/rtsp-server/rtsp-media.c:
7994 media: only flush when setting a new start position
7995 Only flush the pipeline when we change the start position with
7997 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7999 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
8001 * gst/rtsp-server/rtsp-stream.c:
8002 stream: set ttl-mc before adding the socket
8003 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
8004 never be set on socket.
8005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
8007 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8009 * gst/rtsp-server/rtsp-media.c:
8010 media: stop thread if media is already prepared
8011 in gst_rtsp_media_prepare() the thread is not used if media is already
8012 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
8014 https://bugzilla.gnome.org/show_bug.cgi?id=724182
8016 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
8019 build: Ship gst-rtsp-server.doap file
8021 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
8023 * tests/check/gst/rtspserver.c:
8024 tests: Fix another compiler warning with gcc
8026 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
8028 * gst/rtsp-server/rtsp-client.c:
8029 * gst/rtsp-server/rtsp-mount-points.c:
8030 * gst/rtsp-server/rtsp-stream.c:
8031 * tests/check/gst/client.c:
8032 rtsp-server: Fix lots of compiler warnings with clang
8034 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
8037 * gst-rtsp-server.doap:
8038 * tests/Makefile.am:
8039 configure: Synchronise with the configure scripts of the other modules
8041 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
8044 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
8046 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
8048 * gst/rtsp-server/rtsp-media.c:
8049 * gst/rtsp-server/rtsp-stream.c:
8050 Revert "rtsp-server: support build against last stable release"
8051 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
8052 Let us require 1.2.3 now, which is going to be released in a few
8055 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
8057 * gst/rtsp-server/rtsp-session-media.c:
8058 * gst/rtsp-server/rtsp-stream-transport.c:
8059 session: improve RTP-Info
8060 Ignore streams that can't generate RTP-Info instead of failing.
8061 Don't return the empty string when all streams are unconfigured but
8062 return NULL so that we don't generate and empty RTP-Info header.
8063 Improve docs a little.
8065 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
8067 * gst/rtsp-server/rtsp-session-media.c:
8068 Don't free rtpinfo GString when it is NULL
8069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8071 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
8073 * gst/rtsp-server/rtsp-media.c:
8074 media: only set keyframe flag when modifying start
8075 Only set the keyframe flag when we modify the start position. The
8076 keyframe flag should probably be ignored when no change is requested but
8077 until we can claim this is all documented properly and all demuxer
8078 implement this, avoid setting the flag.
8079 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
8081 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
8083 * gst/rtsp-server/rtsp-thread-pool.c:
8084 thread-pool: Unref source after mainloop has quit to avoid races in GLib
8085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
8087 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
8089 * gst/rtsp-server/rtsp-stream.c:
8090 stream: handle NULL seqnum and rtptime arguments
8092 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
8094 * gst/rtsp-server/rtsp-thread-pool.c:
8095 * tests/check/gst/threadpool.c:
8096 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
8097 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
8099 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
8101 * gst/rtsp-server/rtsp-stream.c:
8102 stream: add fallback for missing stats property
8103 Use a fallback when the payloader does not have a stats property
8104 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8106 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
8109 Automatic update of common submodule
8110 From f7bc1c3 to 1a07da9
8112 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
8114 * gst/rtsp-server/rtsp-stream.c:
8115 stream: don't leak stats structure
8116 Don't leak the stats structure and deal with NULL stats.
8118 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
8120 * gst/rtsp-server/rtsp-stream.c:
8121 stream: Get rtpinfo properties atomically from payloader
8122 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
8124 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
8126 * gst/rtsp-server/rtsp-media.c:
8127 media: refactor state change functions and signals
8128 Make functions to set the target state and the pipeline state and emit
8129 the signals from those functions.
8131 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
8133 * gst/rtsp-server/rtsp-media.c:
8134 * gst/rtsp-server/rtsp-media.h:
8135 media: add signal to notify of pending state changes
8137 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
8139 * gst/rtsp-server/rtsp-media.c:
8140 * gst/rtsp-server/rtsp-stream.c:
8141 rtsp-server: support build against last stable release
8142 Until 1.2.3 is out with the new get_type function and we
8145 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
8147 * gst/rtsp-server/rtsp-stream.c:
8148 stream: fix compilation
8150 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
8152 * gst/rtsp-server/rtsp-media.c:
8153 * gst/rtsp-server/rtsp-media.h:
8154 * gst/rtsp-server/rtsp-stream.c:
8155 * gst/rtsp-server/rtsp-stream.h:
8156 stream: add property to configure profiles
8158 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
8160 * gst/rtsp-server/rtsp-client.c:
8161 client: let stream check supported transport
8162 Delegate the check if a transport is allowed to the stream.
8163 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
8165 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
8167 * gst/rtsp-server/rtsp-stream.c:
8168 * gst/rtsp-server/rtsp-stream.h:
8169 stream: add method to check supported transport
8170 Add a method to check if a transport is supported
8172 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
8175 configure.ac: Only check for gstreamer-check, not check
8176 We include check in gstreamer-check since quite some time now.
8178 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
8180 * gst/rtsp-server/rtsp-session-media.c:
8181 * gst/rtsp-server/rtsp-stream-transport.c:
8182 * gst/rtsp-server/rtsp-stream.c:
8183 * gst/rtsp-server/rtsp-stream.h:
8184 stream: return clock-rate from get_rtpinfo
8185 And use it to correct the rtptime to the requested start-time.
8186 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
8188 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
8190 * gst/rtsp-server/rtsp-session-media.c:
8191 * gst/rtsp-server/rtsp-stream-transport.c:
8192 * gst/rtsp-server/rtsp-stream-transport.h:
8193 session-media: calculate start-time
8195 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
8197 * gst/rtsp-server/rtsp-stream-transport.c:
8198 * gst/rtsp-server/rtsp-stream.c:
8199 * gst/rtsp-server/rtsp-stream.h:
8200 stream: also return the running-time
8201 Return the running-time in the rtpinfo as well.
8203 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
8205 * gst/rtsp-server/rtsp-client.c:
8206 * gst/rtsp-server/rtsp-session-media.c:
8207 * gst/rtsp-server/rtsp-session-media.h:
8208 * gst/rtsp-server/rtsp-stream-transport.c:
8209 * gst/rtsp-server/rtsp-stream-transport.h:
8210 session-media: let the session-media make the RTPInfo
8211 Add method to create the RTPInfo for a stream-transport.
8212 Add method to create the RTPInfo for all stream-transports in a
8214 Use the session-media RTPInfo code in client. This allows us to refactor
8215 another method to link the TCP callbacks.
8217 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8219 mount-points: sort sequence before g_sequence_lookup
8220 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
8221 sort sequence if dirty, otherwise lookup will fail.
8222 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
8224 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
8227 configure: rename package from gst-rtsp to gst-rtsp-server
8228 To match git module name and avoid confusion with the
8229 rtsp lib in gst-plugins-base and rtsp plugin in -good.
8231 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
8234 configure: bump core/base/good requirement to 1.2.0
8235 Bump to released stable version and make implicit
8236 requirements explicit.
8238 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
8243 Fix broken gettext setup which is not used anyway
8245 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
8248 Automatic update of common submodule
8249 From dbedaa0 to d48bed3
8251 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
8253 * gst/rtsp-server/rtsp-client.c:
8254 * gst/rtsp-server/rtsp-media.c:
8255 * gst/rtsp-server/rtsp-media.h:
8256 media: add setup_sdp vmethod
8257 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
8258 gst_rtsp_media_setup_sdp.
8259 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
8261 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
8263 * gst/rtsp-server/rtsp-stream.c:
8264 rtsp-stream: Check return value of sscanf
8265 streamid is only valid if sscanf matched something.
8267 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
8269 * gst/rtsp-server/rtsp-client.c:
8270 rtsp-client: Fix iteration
8271 Wouldn't even enter the code block otherwise (i++ was used as the check
8272 and not the postfix).
8274 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
8276 * gst/rtsp-server/rtsp-client.c:
8277 * gst/rtsp-server/rtsp-client.h:
8278 client: add vmethod to configure media and streams
8279 Implement a vmethod that can be used to configure the media and the
8280 streams based on the current context. Handle the blocksize handling in
8281 the default handler.
8282 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
8284 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8287 Make git ignore more unit test binaries
8289 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8291 * gst/rtsp-server/rtsp-address-pool.h:
8292 * gst/rtsp-server/rtsp-auth.h:
8293 * gst/rtsp-server/rtsp-client.h:
8294 * gst/rtsp-server/rtsp-context.h:
8295 * gst/rtsp-server/rtsp-media-factory-uri.h:
8296 * gst/rtsp-server/rtsp-media-factory.h:
8297 * gst/rtsp-server/rtsp-media.h:
8298 * gst/rtsp-server/rtsp-mount-points.h:
8299 * gst/rtsp-server/rtsp-server.h:
8300 * gst/rtsp-server/rtsp-session-media.h:
8301 * gst/rtsp-server/rtsp-session-pool.h:
8302 * gst/rtsp-server/rtsp-session.h:
8303 * gst/rtsp-server/rtsp-stream-transport.h:
8304 * gst/rtsp-server/rtsp-stream.h:
8305 * gst/rtsp-server/rtsp-thread-pool.h:
8306 * gst/rtsp-server/rtsp-token.h:
8307 rtsp-server: add padding to many public structures
8308 Not mini objects though, since they are not subclassable
8309 anyway, nor kept on the stack or inlined in a structure.
8311 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8313 media: add new create_rtpbin vmethod
8314 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8315 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8317 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8319 * tests/check/gst/media.c:
8320 tests: fix memory leak, free test's thread pool
8321 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8323 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8325 * gst/rtsp-server/rtsp-stream-transport.c:
8326 stream-transport: free url in finalize
8328 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8330 * gst/rtsp-server/rtsp-media.c:
8331 media: also do state change in suspended state
8333 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8335 * gst/rtsp-server/rtsp-client.c:
8336 * gst/rtsp-server/rtsp-media.c:
8337 media: also handle prepare and range in suspended state
8338 When we are suspended, we are already prepared.
8339 We can get the range in the suspended state.
8341 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8343 * tests/check/Makefile.am:
8344 * tests/check/gst/sessionmedia.c:
8345 check: add test for uri in setup
8346 Added unit tests for the new functionality in GstRTSPStreamTransport.
8347 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8349 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8351 * gst/rtsp-server/rtsp-client.c:
8352 client: store setup uri and use in PLAY response
8353 Store the uri used when doing the setup and use that in the PLAY
8355 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8357 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8359 * gst/rtsp-server/rtsp-stream-transport.c:
8360 * gst/rtsp-server/rtsp-stream-transport.h:
8361 stream-transport: add method to get/set url
8363 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8365 * gst/rtsp-server/rtsp-client.c:
8366 client: suspend after SDP and unsuspend before PLAYING
8367 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8368 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8370 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8372 * gst/rtsp-server/rtsp-media-factory.c:
8373 * gst/rtsp-server/rtsp-media-factory.h:
8374 * gst/rtsp-server/rtsp-media.c:
8375 * gst/rtsp-server/rtsp-media.h:
8376 * gst/rtsp-server/rtsp-session-media.c:
8377 * gst/rtsp-server/rtsp-session.c:
8378 * tests/check/gst/media.c:
8379 * tests/check/gst/mediafactory.c:
8380 media: add suspend modes
8381 Add support for different suspend modes. The stream is suspended right after
8382 producing the SDP and after PAUSE. Different suspend modes are available that
8383 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8384 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8385 state and RESET will bring the pipeline to the NULL state.
8386 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8387 this means that the pipeline needs to be prerolled again.
8388 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8389 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8391 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8393 * gst/rtsp-server/rtsp-media.c:
8394 media: start live streams in blocked state
8395 Start live streams in the blocked state and make them preroll using the
8396 messages. This ensure that no data is played by the sink until we explicitly
8397 unblock the stream right before going to PLAYING.
8398 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8400 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8402 * gst/rtsp-server/rtsp-media.c:
8403 media: refactor starting and waiting for preroll
8404 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8405 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8407 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8409 * gst/rtsp-server/rtsp-stream.c:
8410 * gst/rtsp-server/rtsp-stream.h:
8411 stream: add API to block streams
8412 Add an API to block on the streams and make it post a message.
8413 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8414 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8416 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8418 * docs/libs/Makefile.am:
8419 docs: Specify the override file
8420 Even if it's empty (for now) it avoids make distcheck complaining
8422 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8424 * gst/rtsp-server/rtsp-media.c:
8425 media: move default implementations to where they are used
8427 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8429 * gst/rtsp-server/rtsp-media.c:
8430 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8431 We need to take the state_lock when calling this method.
8433 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8435 * gst/rtsp-server/rtsp-media.c:
8436 media: handle add-added on non-bins too
8437 Handle dynamic payloaders that are not bins, as used in the unit-test.
8439 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8441 * gst/rtsp-server/rtsp-media-factory.c:
8442 * gst/rtsp-server/rtsp-media-factory.h:
8443 * gst/rtsp-server/rtsp-media.c:
8444 rtsp-media/-factory: Fix request pad name comments
8445 These must be escaped for gtk-doc to parse the comments without warnings.
8447 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8449 rtsp-media: remove transports if media is in error status
8450 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8451 trying to change to GST_STATE_NULL and media is in error status, we
8452 remove all transports.
8453 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8455 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8457 * gst/rtsp-server/rtsp-media.c:
8458 rtsp-media: use element metadata to find payloader
8459 Use the element metadata to find the payloader instead of checking
8461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8463 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8465 rtsp-stream: add getter for payload type
8466 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8467 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8468 element and create the stream with this one instead of the dynpay%d
8470 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8472 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8474 * gst/rtsp-server/rtsp-client.c:
8475 * gst/rtsp-server/rtsp-context.h:
8476 * gst/rtsp-server/rtsp-media.c:
8477 * gst/rtsp-server/rtsp-mount-points.c:
8478 * gst/rtsp-server/rtsp-server.c:
8479 * gst/rtsp-server/rtsp-token.c:
8480 rtsp-*: Refer to NULL as a constant in comments
8482 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8484 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8486 rtsp-*: Fix type name typos in comments
8487 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8488 * rtsp-auth: Refer to part of constant name as text
8489 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8490 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8491 * rtsp-stream: Fix typo when refering to GstBin
8492 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8494 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8497 * docs/libs/gst-rtsp-server-docs.sgml:
8498 * docs/libs/gst-rtsp-server-sections.txt:
8499 docs: Improve documentation
8500 * Include annotation-glossary to quiet gtk-doc
8501 * Rename remaining ClientState -> Context
8502 * Rename object hierarchy file
8503 * Remove stale chapter references
8504 * Add missing function and object references
8505 * Include missing GstRTSPAddressPoolResult
8506 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8508 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8510 * gst/rtsp-server/rtsp-client.c:
8511 * gst/rtsp-server/rtsp-server.c:
8512 * gst/rtsp-server/rtsp-session-pool.c:
8513 * gst/rtsp-server/rtsp-session.c:
8514 * gst/rtsp-server/rtsp-stream.c:
8515 rtsp-server: sprinkle some allow-none annotations for g-i
8517 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8519 * gst/rtsp-server/rtsp-stream.c:
8520 * gst/rtsp-server/rtsp-stream.h:
8521 stream: add method to filter transports
8522 Add a method to safely iterate and collect the stream transports
8523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8525 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8527 * gst/rtsp-server/rtsp-client.c:
8528 * gst/rtsp-server/rtsp-server.c:
8529 * gst/rtsp-server/rtsp-session-pool.c:
8530 * gst/rtsp-server/rtsp-session.c:
8531 rtsp: allow NULL func in filters
8532 Passing a null function make the filters return a list of
8535 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8537 * gst/rtsp-server/rtsp-address-pool.c:
8538 * tests/check/gst/addresspool.c:
8539 address-pool: fix address increment
8540 Use a guint instead of guint8 to increment the address. It's still not
8541 completely correct because a guint might not be able to hold the complete
8542 address range, but that's an enhacement for later.
8543 Add unit test to test improved behaviour.
8544 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8546 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8548 * gst/rtsp-server/rtsp-client.c:
8549 * tests/check/gst/client.c:
8550 client: allow absolute path in requests
8551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8553 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8555 * gst/rtsp-server/rtsp-client.c:
8556 * gst/rtsp-server/rtsp-client.h:
8557 client: make make_path_from_uri a vmethod
8559 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8561 * docs/libs/gst-rtsp-server-sections.txt:
8562 * gst/rtsp-server/rtsp-stream.c:
8563 * gst/rtsp-server/rtsp-stream.h:
8564 * tests/check/Makefile.am:
8565 * tests/check/gst/stream.c:
8566 stream: Add functions to get rtp and rtcp sockets
8567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8569 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8571 * gst/rtsp-server/rtsp-context.c:
8572 * gst/rtsp-server/rtsp-context.h:
8573 context: defing a GType for the context
8574 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8576 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8578 * gst/rtsp-server/Makefile.am:
8579 * gst/rtsp-server/rtsp-auth.c:
8580 * gst/rtsp-server/rtsp-context.c:
8581 * gst/rtsp-server/rtsp-media.c:
8582 * gst/rtsp-server/rtsp-mount-points.c:
8583 * gst/rtsp-server/rtsp-server.h:
8584 * gst/rtsp-server/rtsp-session-media.c:
8585 * gst/rtsp-server/rtsp-session.c:
8586 * gst/rtsp-server/rtsp-stream.c:
8587 Fixed several GIR warnings
8589 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8591 * gst/rtsp-server/rtsp-auth.c:
8594 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8596 * tests/check/Makefile.am:
8597 * tests/check/gst/token.c:
8598 tests: Add unit tests for token
8599 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8601 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8603 * gst/rtsp-server/rtsp-token.c:
8604 token: Validate args for gst_rtsp_token_is_allowed
8605 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8607 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8609 * gst/rtsp-server/rtsp-token.c:
8610 token: Fix bug when creating empty token
8611 We always want to have a valid GstStructure in the token.
8612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8614 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8616 * gst/rtsp-server/rtsp-thread-pool.c:
8617 thread-pool: avoid race in shutdown
8618 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8619 don't actually stop the mainloop ever. Solve this race by adding an idle source
8620 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8621 if quit was called before we started it.
8623 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8625 * tests/check/Makefile.am:
8626 * tests/check/gst/permissions.c:
8627 tests: Add unit tests for permissions
8628 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8630 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8632 * tests/check/gst/mediafactory.c:
8633 tests: Test mediafactory permissions
8634 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8636 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8638 * gst/rtsp-server/rtsp-permissions.c:
8639 permissions: Fix refcounting when adding/removing roles
8640 Previously a role that was removed was unreffed twice, and when
8641 replacing an existing role the replaced role was freed while still being
8642 referenced. Both bugs are now fixed.
8643 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8645 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8647 * tests/check/gst/media.c:
8648 * tests/check/gst/mediafactory.c:
8649 * tests/check/gst/rtspserver.c:
8650 tests: Check gst_rtsp_url_parse return value
8651 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8653 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8656 Automatic update of common submodule
8657 From 865aa20 to dbedaa0
8659 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8661 * gst/rtsp-server/rtsp-server.c:
8662 rtsp-server: Fix socket leak
8663 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8665 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8667 * gst/rtsp-server/rtsp-session-pool.c:
8668 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8669 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8671 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8673 * examples/.gitignore:
8674 * examples/test-video.c:
8675 examples: fix compilation when WITH_AUTH is defined
8676 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8678 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8681 gitignore: Add new test binary
8683 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8685 * tests/check/Makefile.am:
8686 * tests/check/gst/threadpool.c:
8687 thread-pool: Add unit test for the thread pools
8688 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8690 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8692 * gst/rtsp-server/rtsp-thread-pool.c:
8693 thread-pool: Fix thread leak when reusing threads
8694 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8696 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8698 * gst/rtsp-server/rtsp-server.c:
8699 * tests/check/gst/rtspserver.c:
8700 tests: fixed racy behavior in rtspserver tests
8701 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8703 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8705 * tests/check/gst/addresspool.c:
8706 tests: Improve address pool unit tests
8707 Add a range with mixed IPV4 and IPV6 addresses to pool.
8708 Get an IPV4 address from an IPV6-only pool.
8709 Get an IPV6 address from an IPV4-only pool.
8710 Reserve a IPV6 address from an IPV4-only pool.
8711 Check for unicast addresses in multicast-only pool.
8712 Check for unicast addresses in uni-/multicast-mixed pool.
8713 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8715 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8717 * gst/rtsp-server/rtsp-client.c:
8718 client: append query string in PAUSE/PLAY/TEARDOWN as well
8720 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8722 * gst/rtsp-server/rtsp-client.c:
8723 client: Add query to control path
8724 If the SETUP url contains a query it must be appended to the control
8725 path so that it matches any already created stream in the media. The
8726 query will also be appended to the session media path.
8728 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8730 * gst/rtsp-server/rtsp-media.c:
8731 rtsp-media: remove old line
8733 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8735 * gst/rtsp-server/rtsp-stream.c:
8736 stream: Correct control comparison
8737 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8739 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8741 * gst/rtsp-server/rtsp-media.c:
8742 media: Check dynamically if the pipeline supports seeking
8743 We should not depend on whether or not the pipeline state change
8744 returned NO_PREROLL or not. A media could dynamically change its
8745 element and switch from seekable to non seekable so it's best to test
8746 the seekable nature of the pipeline dynamically when we try to do a seek.
8748 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8750 * gst/rtsp-server/rtsp-media.c:
8751 media: Return FALSE if seeking is not supported
8753 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8755 * gst/rtsp-server/rtsp-media.c:
8756 rtsp-media: don't seek accurate by default
8757 Accurate seeking is perhaps a little overkill in the most common situation and
8758 causes some formats (mp3) over slow media to seek extremely slowly.
8760 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8762 * tests/check/gst/rtspserver.c:
8763 tests: fix unit test
8764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8766 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8768 * gst/rtsp-server/rtsp-client.c:
8769 client: Reply 400 if media cannot be constructed
8770 Reply 400 Bad Request instead of 503 Service Unavailable if media
8771 cannot be constructed in SETUP.
8772 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8774 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8776 * gst/rtsp-server/rtsp-client.c:
8777 client: Send setup reply once only
8778 If find_media() failed in handle_setup_request() two replies was sent.
8779 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8781 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8784 Automatic update of common submodule
8785 From 6b03ba7 to 865aa20
8787 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8789 * gst/rtsp-server/rtsp-server.c:
8790 server: Emit client-connected signal earlier
8791 Emit client-connected before the client ref is given to a GSource,
8792 otherwise client-connected can be emitted after the client object has
8795 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8797 * gst/rtsp-server/rtsp-address-pool.c:
8798 * gst/rtsp-server/rtsp-address-pool.h:
8799 * gst/rtsp-server/rtsp-stream.c:
8800 * tests/check/gst/addresspool.c:
8801 addresspool: return reason of failure
8802 Let gst_rtsp_address_pool_reserve_address() return the reason why
8803 the address could not be reserved.
8804 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8806 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8809 autogen.sh: Sync behaviour with other GStreamer modules
8810 Allows building from outside of tree amongst other things
8812 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8815 Automatic update of common submodule
8816 From b613661 to 6b03ba7
8818 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8821 Automatic update of common submodule
8822 From 74a6857 to b613661
8824 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8827 Automatic update of common submodule
8828 From 01a7a46 to 74a6857
8830 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8832 * gst/rtsp-server/rtsp-client.c:
8833 client: Do not read beyond end of path string
8834 If the setup was done without a control url, make sure we don't try to read the
8835 non-existing control string and crash.
8837 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8839 * gst/rtsp-server/rtsp-client.c:
8840 client: Fix RTPInfo header
8841 Refactor the method to make the content_base.
8842 Use the content-base and the control url to construct the RTPInfo
8845 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8847 * gst/rtsp-server/rtsp-client.c:
8848 client: map url to path only in describe
8849 Only map the request url to a path in the DESCRIBE method. The SDP then
8850 contains the base and control urls that should be used to SETUP/PAUSE/
8851 PLAY/TEARDOWN the media.
8853 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8855 * gst/rtsp-server/rtsp-client.c:
8856 Revert "client: map URL to path in requests"
8857 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8858 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8859 contains the base and control urls which are used in the SETUP, PLAY,
8860 PAUSE and TEARDOWN requests.
8862 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8864 * gst/rtsp-server/rtsp-client.c:
8865 client: map URL to path in requests
8867 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8869 * gst/rtsp-server/rtsp-client.c:
8870 * gst/rtsp-server/rtsp-mount-points.c:
8871 * gst/rtsp-server/rtsp-mount-points.h:
8872 mount-points: make vmethod to make path from uri
8873 Make a vmethod to transform an url into a path. The path is then used to lookup
8874 the factory. This makes it possible to also use other bits of the url, such as
8875 the query parameters, to locate the factory.
8877 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8879 * gst/rtsp-server/rtsp-thread-pool.c:
8880 * gst/rtsp-server/rtsp-thread-pool.h:
8881 thread-pool: Add cleanup to wait for the threadpool to finish
8882 Also fix race condition if two threads are asking for the first
8883 thread from the thread pool at once. This would case two internal
8884 GThreadPools to be created.
8885 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8887 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8889 * gst/rtsp-server/rtsp-client.c:
8890 * tests/check/gst/client.c:
8891 client: free threadpool
8892 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8894 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8896 * tests/check/gst/mountpoints.c:
8897 mountpoints tests: unref matched factories
8898 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8900 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8902 * tests/check/gst/media.c:
8903 media tests: unref thread pool and caps
8904 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8906 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8908 * gst/rtsp-server/rtsp-auth.c:
8909 * gst/rtsp-server/rtsp-media-factory.c:
8910 * gst/rtsp-server/rtsp-media.c:
8911 auth, media, media-factory: unref permissions
8912 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8914 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8916 * examples/Makefile.am:
8917 Makefile: add rule for appsrc example
8919 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8921 * examples/test-appsrc.c:
8922 tests: add appsrc example
8923 Add an example on how to use appsrc to feed the server pipeline with data.
8925 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8927 * gst/rtsp-server/rtsp-client.c:
8928 rtsp-client: remove query part from content-base string
8929 Make sure that after the control url has been resolved, it's
8930 not a part of the query-string.
8931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8933 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8935 * gst/rtsp-server/rtsp-client.c:
8936 client: don't check url in response
8937 There is no url or method in the response to check
8939 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8941 * gst/rtsp-server/rtsp-client.c:
8942 * gst/rtsp-server/rtsp-client.h:
8943 Add handle-response signal for when we receive a GET_PARAMETER response
8945 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8947 * gst/rtsp-server/rtsp-server.c:
8948 Fix gst_rtsp_server_client_filter, using wrong variable type
8950 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8952 * gst/rtsp-server/rtsp-media-factory-uri.c:
8953 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8954 For AAC we need to check for framed=true instead of parsed=true.
8955 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8957 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8959 * gst/rtsp-server/rtsp-stream.c:
8960 stream: optimize pipeline for protocols
8961 When TCP is not an allowed protocol for the stream, avoid creating the
8962 appsrc/appsink/queue and tee elements.
8964 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8966 * gst/rtsp-server/rtsp-media.c:
8967 media: set protocols on streams
8969 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8971 * gst/rtsp-server/rtsp-client.c:
8972 client: use protocols supported by stream
8974 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8976 * gst/rtsp-server/rtsp-media-factory.c:
8977 * gst/rtsp-server/rtsp-media.c:
8978 * gst/rtsp-server/rtsp-stream.c:
8979 media-factory: allow all protocols
8981 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8983 * gst/rtsp-server/rtsp-media.c:
8984 media: configure protocols in new streams
8986 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8988 * gst/rtsp-server/rtsp-stream.c:
8989 * gst/rtsp-server/rtsp-stream.h:
8990 stream: add protocols property
8992 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8994 * gst/rtsp-server/rtsp-media.c:
8995 rtsp-media: send state in "new-state" signal
8996 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8998 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
9001 build: add subdir-objects to AM_INIT_AUTOMAKE
9002 Fixes warnings with automake 1.14
9003 https://bugzilla.gnome.org/show_bug.cgi?id=705350
9005 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9007 * docs/libs/gst-rtsp-server-sections.txt:
9008 * gst/rtsp-server/rtsp-client.c:
9009 * gst/rtsp-server/rtsp-server.c:
9010 * gst/rtsp-server/rtsp-server.h:
9011 server: add method to iterate clients of server
9013 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9015 * gst/rtsp-server/rtsp-media.c:
9016 * gst/rtsp-server/rtsp-media.h:
9017 Add vmethod for rtsp-media subclass to access rtpbin
9019 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9021 * gst/rtsp-server/rtsp-client.h:
9022 small documentation fix
9024 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9026 * gst/rtsp-server/rtsp-client.c:
9027 Do not take range header if range is invalid
9029 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9031 * docs/libs/gst-rtsp-server-sections.txt:
9032 * gst/rtsp-server/rtsp-media.c:
9033 media: add docs for new method
9035 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9037 * gst/rtsp-server/rtsp-media.c:
9038 * gst/rtsp-server/rtsp-media.h:
9039 Add API to rtsp-media set the pipeline's state
9041 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9043 * gst/rtsp-server/rtsp-media.c:
9044 Update current position/duration when gst_rtsp_media_get_range_string is called
9046 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9048 * examples/test-cgroups.c:
9049 tests: add some more docs
9051 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9053 * examples/test-cgroups.c:
9054 * gst/rtsp-server/Makefile.am:
9055 * gst/rtsp-server/rtsp-auth.c:
9056 * gst/rtsp-server/rtsp-auth.h:
9057 * gst/rtsp-server/rtsp-client.c:
9058 * gst/rtsp-server/rtsp-client.h:
9059 * gst/rtsp-server/rtsp-context.c:
9060 * gst/rtsp-server/rtsp-context.h:
9061 * gst/rtsp-server/rtsp-params.c:
9062 * gst/rtsp-server/rtsp-params.h:
9063 * gst/rtsp-server/rtsp-server.c:
9064 * gst/rtsp-server/rtsp-thread-pool.c:
9065 * gst/rtsp-server/rtsp-thread-pool.h:
9066 * tests/check/gst/client.c:
9067 ClientState -> Context
9068 Rename the clientstate to context and put the code in a separate file.
9070 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9072 * examples/test-auth.c:
9073 * gst/rtsp-server/rtsp-auth.c:
9074 * gst/rtsp-server/rtsp-auth.h:
9075 auth: add support for default token
9076 The default token is used when the user is not authenticated and can be used to
9077 give minimal permissions.
9079 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9081 * examples/test-auth.c:
9082 * gst/rtsp-server/rtsp-auth.c:
9083 auth: use defines when possible
9085 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9087 * gst/rtsp-server/rtsp-address-pool.c:
9088 address-pool: improve docs
9090 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9092 * gst/rtsp-server/rtsp-permissions.c:
9093 permissions: add the role to the copy
9095 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
9097 * gst/rtsp-server/rtsp-permissions.c:
9098 permissions: Also copy the roles
9100 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
9102 * gst/rtsp-server/rtsp-permissions.c:
9103 permissions: Make it build
9105 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-address-pool.h:
9110 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * docs/libs/gst-rtsp-server-sections.txt:
9113 * gst/rtsp-server/rtsp-auth.c:
9114 * gst/rtsp-server/rtsp-auth.h:
9115 * gst/rtsp-server/rtsp-media.c:
9116 * gst/rtsp-server/rtsp-session-media.c:
9117 * gst/rtsp-server/rtsp-stream-transport.c:
9118 * gst/rtsp-server/rtsp-stream-transport.h:
9119 * gst/rtsp-server/rtsp-stream.c:
9120 * tests/check/gst/client.c:
9123 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * docs/libs/gst-rtsp-server-sections.txt:
9126 * gst/rtsp-server/rtsp-address-pool.c:
9127 * gst/rtsp-server/rtsp-address-pool.h:
9128 * tests/check/gst/addresspool.c:
9129 * tests/check/gst/rtspserver.c:
9130 address-pool: cleanups
9131 Remove redundant method, improve docs.
9133 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9135 * docs/libs/gst-rtsp-server-sections.txt:
9136 * gst/rtsp-server/rtsp-auth.h:
9137 * gst/rtsp-server/rtsp-permissions.c:
9138 * gst/rtsp-server/rtsp-permissions.h:
9139 * gst/rtsp-server/rtsp-token.c:
9142 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9144 * gst/rtsp-server/rtsp-permissions.c:
9145 permissions: implement _remove_role
9147 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9149 * gst/rtsp-server/rtsp-permissions.c:
9150 permissions: update docs
9152 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9154 * tests/check/gst/client.c:
9155 tests: simplify tests
9156 Client settings are now disabled by default so we don't need an auth
9157 module to disable them.
9159 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9161 * gst/rtsp-server/rtsp-auth.c:
9162 auth: add default authorizations
9163 When no auth module is specified, use our table of defaults to look up the
9164 default value of the check instead of always allowing everything. This was
9165 we can disallow client settings by default.
9167 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9170 README: update readme
9172 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9174 * gst/rtsp-server/rtsp-thread-pool.c:
9175 * gst/rtsp-server/rtsp-thread-pool.h:
9176 thread-pool: add more docs
9178 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9180 * gst/rtsp-server/rtsp-thread-pool.c:
9181 * gst/rtsp-server/rtsp-thread-pool.h:
9182 thread-pool: fix race in thread reuse
9183 If we try to reuse a thread right after we made it stop, we end up using a
9184 stopped thread. Catch this case and only reuse threads that are not stopping.
9186 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9188 * gst/rtsp-server/rtsp-server.c:
9189 server: add small debug
9191 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9193 * tests/check/gst/client.c:
9195 Add some permissions to media so we can use the auth and enable
9198 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9200 * gst/rtsp-server/rtsp-client.c:
9201 client: support pushed context in handle_request
9202 If we already have a pushed state, reuse it and add our own things. This makes
9203 it easier to write tests.
9205 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9207 * gst/rtsp-server/rtsp-auth.c:
9208 auth: don't auth on methods
9209 Don't authorize on methods anymore but on the resources that we
9210 try to access, this is more flexible.
9211 Move the authorization checks to where they are needed and let the
9212 check return the response on error.
9214 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9216 * gst/rtsp-server/rtsp-mount-points.c:
9217 mount-points: add some debug
9219 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9221 * tests/check/gst/client.c:
9222 tests: almost fix test
9224 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9226 * gst/rtsp-server/rtsp-auth.c:
9227 * gst/rtsp-server/rtsp-auth.h:
9228 * gst/rtsp-server/rtsp-client.c:
9229 * gst/rtsp-server/rtsp-client.h:
9230 * gst/rtsp-server/rtsp-server.c:
9231 * gst/rtsp-server/rtsp-server.h:
9232 auth: let the auth module check client_settings
9233 Let the auth module decide if client settings are allowed for the
9236 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9238 * gst/rtsp-server/rtsp-token.c:
9239 * gst/rtsp-server/rtsp-token.h:
9240 token: add method to check boolean permission
9242 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9244 * examples/test-auth.c:
9245 * examples/test-cgroups.c:
9246 * gst/rtsp-server/rtsp-token.c:
9247 * gst/rtsp-server/rtsp-token.h:
9248 token: simplify token constructor
9249 Use variable arguments to make easier API.
9251 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9253 * examples/test-auth.c:
9254 * examples/test-cgroups.c:
9255 * gst/rtsp-server/rtsp-media-factory.c:
9256 * gst/rtsp-server/rtsp-media-factory.h:
9257 media-factory: add convenience API for factory
9259 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9261 * examples/test-auth.c:
9262 * examples/test-cgroups.c:
9263 * gst/rtsp-server/rtsp-permissions.c:
9264 * gst/rtsp-server/rtsp-permissions.h:
9265 permissions: simplify API a little
9266 Avoid passing GstStructure in the add_role method, use varargs instead
9267 to construct the structure behind the scenes. We can then also use the
9268 structure name as the role and simplify some more logic.
9270 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9272 * gst/rtsp-server/rtsp-auth.c:
9275 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9277 * gst/rtsp-server/rtsp-auth.c:
9278 * gst/rtsp-server/rtsp-auth.h:
9279 * gst/rtsp-server/rtsp-client.c:
9280 auth: handle unauthorized response
9281 Move handling of the unauthorized response to the auth module, it can add
9282 the appropriate headers to request authorization for the required method
9283 much better than the client.
9285 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9287 * gst/rtsp-server/rtsp-client.c:
9288 * gst/rtsp-server/rtsp-client.h:
9289 client: allow for sending any message, not only requests
9290 Change the _send_request() method to _send_message() so that we
9291 can both send requests and replies.
9293 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9295 * docs/libs/gst-rtsp-server-sections.txt:
9296 * gst/rtsp-server/rtsp-server.h:
9299 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9301 * examples/test-video.c:
9302 * gst/rtsp-server/rtsp-auth.c:
9303 * gst/rtsp-server/rtsp-auth.h:
9304 * gst/rtsp-server/rtsp-server.c:
9305 * gst/rtsp-server/rtsp-server.h:
9306 auth: move TLS handling to auth module
9307 Remove the TLS settings on the server and move it to the auth module because
9308 that is where security related bits go.
9310 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9312 * gst/rtsp-server/rtsp-client.c:
9313 * gst/rtsp-server/rtsp-client.h:
9314 client: add state push/pop
9316 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9318 * gst/rtsp-server/rtsp-client.c:
9319 * gst/rtsp-server/rtsp-client.h:
9320 client: add connection to state
9322 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9324 * gst/rtsp-server/rtsp-mount-points.c:
9325 mount-points: fix debug
9327 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9329 * tests/check/gst/media.c:
9330 tests: fix media test
9332 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9334 * gst/rtsp-server/rtsp-thread-pool.c:
9335 thread-pool: we don't require a state
9337 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9339 * gst/rtsp-server/rtsp-server.c:
9340 server: let context ref the server
9341 So that we don't risk losing the server object early anc crash.
9343 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9345 * tests/check/gst/client.c:
9346 tests: fix client test
9348 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9351 * docs/libs/gst-rtsp-server-docs.sgml:
9352 * docs/libs/gst-rtsp-server-sections.txt:
9353 * gst/rtsp-server/rtsp-address-pool.c:
9354 * gst/rtsp-server/rtsp-auth.c:
9355 * gst/rtsp-server/rtsp-client.c:
9356 * gst/rtsp-server/rtsp-client.h:
9357 * gst/rtsp-server/rtsp-media-factory-uri.c:
9358 * gst/rtsp-server/rtsp-media-factory.c:
9359 * gst/rtsp-server/rtsp-media-factory.h:
9360 * gst/rtsp-server/rtsp-media.c:
9361 * gst/rtsp-server/rtsp-mount-points.c:
9362 * gst/rtsp-server/rtsp-params.c:
9363 * gst/rtsp-server/rtsp-permissions.c:
9364 * gst/rtsp-server/rtsp-sdp.c:
9365 * gst/rtsp-server/rtsp-server.c:
9366 * gst/rtsp-server/rtsp-server.h:
9367 * gst/rtsp-server/rtsp-session-media.c:
9368 * gst/rtsp-server/rtsp-session-pool.c:
9369 * gst/rtsp-server/rtsp-session.c:
9370 * gst/rtsp-server/rtsp-stream-transport.c:
9371 * gst/rtsp-server/rtsp-stream.c:
9372 * gst/rtsp-server/rtsp-thread-pool.c:
9373 * gst/rtsp-server/rtsp-token.c:
9376 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9378 * gst/rtsp-server/rtsp-session-pool.c:
9379 * gst/rtsp-server/rtsp-session-pool.h:
9380 session-pool: make vmethod to create a session
9381 Make a vmethod to create a sessions so that subclasses can create
9382 custom session objects
9384 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9386 * gst/rtsp-server/rtsp-auth.c:
9387 * gst/rtsp-server/rtsp-media-factory.h:
9388 * gst/rtsp-server/rtsp-media.h:
9389 * gst/rtsp-server/rtsp-mount-points.h:
9390 * gst/rtsp-server/rtsp-session-pool.h:
9391 * gst/rtsp-server/rtsp-stream.h:
9394 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9396 * docs/libs/gst-rtsp-server-docs.sgml:
9397 * docs/libs/gst-rtsp-server-sections.txt:
9398 * gst/rtsp-server/rtsp-address-pool.c:
9399 * gst/rtsp-server/rtsp-address-pool.h:
9400 * gst/rtsp-server/rtsp-auth.c:
9401 * gst/rtsp-server/rtsp-client.h:
9402 * gst/rtsp-server/rtsp-media-factory.h:
9403 * gst/rtsp-server/rtsp-media.c:
9404 * gst/rtsp-server/rtsp-media.h:
9405 * gst/rtsp-server/rtsp-permissions.c:
9406 * gst/rtsp-server/rtsp-permissions.h:
9407 * gst/rtsp-server/rtsp-server.h:
9408 * gst/rtsp-server/rtsp-session-media.c:
9409 * gst/rtsp-server/rtsp-session-media.h:
9410 * gst/rtsp-server/rtsp-session-pool.h:
9411 * gst/rtsp-server/rtsp-session.h:
9412 * gst/rtsp-server/rtsp-stream-transport.h:
9413 * gst/rtsp-server/rtsp-stream.c:
9414 * gst/rtsp-server/rtsp-thread-pool.h:
9417 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9420 * examples/Makefile.am:
9421 configure: compile cgroup example conditionally
9422 Only compile the cgroup example when we have libcgroup
9424 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * examples/Makefile.am:
9428 * examples/test-cgroups.c:
9429 examples: add cgroups example
9431 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9433 * tests/check/gst/rtspserver.c:
9434 tests: fix compilation
9436 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9438 * gst/rtsp-server/rtsp-thread-pool.c:
9439 thread-pool: fix vmethod invocation
9441 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9443 * gst/rtsp-server/rtsp-thread-pool.c:
9444 * gst/rtsp-server/rtsp-thread-pool.h:
9445 thread-pool: store thread type in thread
9447 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9449 * gst/rtsp-server/rtsp-client.c:
9450 client: pass thread from pool to media _prepare
9451 Get a thread from the configured threadpool and pass it to the prepare method of
9454 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9456 * gst/rtsp-server/rtsp-media.c:
9457 * gst/rtsp-server/rtsp-media.h:
9458 media: Accept a thread in _prepare
9459 Remove out own threadpool handling and use the provided thread and
9460 maincontext for the bus messages and the state changes.
9462 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9464 * gst/rtsp-server/rtsp-server.c:
9465 server: configure client thread pool
9467 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9469 * gst/rtsp-server/rtsp-client.c:
9470 * gst/rtsp-server/rtsp-client.h:
9471 client: add method to configure thread pool
9473 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9475 * gst/rtsp-server/rtsp-client.h:
9476 * gst/rtsp-server/rtsp-server.c:
9477 * gst/rtsp-server/rtsp-server.h:
9478 server: use thread pool
9479 Use the thread pool instead of doing our own thing.
9481 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9483 * gst/rtsp-server/Makefile.am:
9484 * gst/rtsp-server/rtsp-thread-pool.c:
9485 * gst/rtsp-server/rtsp-thread-pool.h:
9486 thread-pool: add object to manage threads
9487 Add an object to manage the client and media threads.
9489 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9491 * gst/rtsp-server/rtsp-auth.c:
9492 auth: debug authorization check
9494 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9496 * gst/rtsp-server/rtsp-media.c:
9497 media: start media pipeline in context
9498 Start the media pipeline in the provided context (or our default one
9499 when NULL). This makes sure that we run the bus thread in this context and that
9500 all media threads are children of this context.
9502 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9504 * gst/rtsp-server/rtsp-media-factory.c:
9505 factory: pass permissions to media by default
9507 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9509 * examples/test-auth.c:
9510 test: add permissions to auth test
9511 Ass some permissions to the media factory in the test.
9513 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9515 * gst/rtsp-server/rtsp-auth.c:
9516 * gst/rtsp-server/rtsp-auth.h:
9517 * gst/rtsp-server/rtsp-client.c:
9518 auth: simplify auth checks
9519 Remove client from methods, it's now in the state
9520 Perform the check specified by the string, use the information from the
9521 thread local context.
9523 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9525 * gst/rtsp-server/rtsp-client.c:
9526 * gst/rtsp-server/rtsp-client.h:
9527 client: add state to current thread
9528 Add the client to the ClientState object.
9529 Place the ClientState on the current thread.
9531 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9533 * gst/rtsp-server/rtsp-media-factory.c:
9534 * gst/rtsp-server/rtsp-media-factory.h:
9535 * gst/rtsp-server/rtsp-media.c:
9536 * gst/rtsp-server/rtsp-media.h:
9537 media: make it possible to set permissions
9538 Make it possible to set permissions on media and media factory objects
9540 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9542 * gst/rtsp-server/Makefile.am:
9543 * gst/rtsp-server/rtsp-permissions.c:
9544 * gst/rtsp-server/rtsp-permissions.h:
9545 permissions: add permissions object
9546 Add a mini object to store permissions based on a role.
9548 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9550 * examples/test-auth.c:
9551 * gst/rtsp-server/rtsp-auth.c:
9552 * gst/rtsp-server/rtsp-auth.h:
9553 * gst/rtsp-server/rtsp-client.c:
9554 auth: add auth checks
9555 Add an enum with auth checks and implement the checks in the auth object.
9556 Perform the checks from the client.
9558 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9560 * examples/test-auth.c:
9561 * gst/rtsp-server/rtsp-auth.c:
9562 * gst/rtsp-server/rtsp-auth.h:
9563 * gst/rtsp-server/rtsp-client.h:
9564 auth: use the token after authentication
9565 After we authenticated a user, keep the Token around in the state.
9567 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9569 * gst/rtsp-server/rtsp-client.c:
9570 * gst/rtsp-server/rtsp-media.c:
9571 * gst/rtsp-server/rtsp-media.h:
9572 * tests/check/gst/media.c:
9573 media: add optional context for bus messages
9574 Add an optional mainloop to _prepare that will handle the bus messages instead
9575 of always using the shared mainloop.
9577 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9579 * gst/rtsp-server/Makefile.am:
9580 * gst/rtsp-server/rtsp-token.c:
9581 * gst/rtsp-server/rtsp-token.h:
9582 token: add authorization token
9583 Add a simply miniobject that contains the authorizations. The object contains a
9584 GstStructure that hold all authorization fields. When a user is authenticated,
9585 the auth module will create a Token for the user. The token is then used to
9586 check what operations the user is allowed to do and various other configuration
9589 2013-07-05 12:08:36 +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.c:
9595 * gst/rtsp-server/rtsp-client.h:
9596 * gst/rtsp-server/rtsp-media-factory.c:
9597 * gst/rtsp-server/rtsp-media-factory.h:
9598 * gst/rtsp-server/rtsp-media.c:
9599 * gst/rtsp-server/rtsp-media.h:
9600 auth: remove auth from media and factory
9601 Remove the auth object from media and factory. We want to have the RTSPClient
9602 authenticate and authorize resources, there is no need to place another auth
9603 manager on the media/factory.
9605 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9607 * examples/test-auth.c:
9608 * gst/rtsp-server/rtsp-auth.c:
9609 * gst/rtsp-server/rtsp-auth.h:
9610 * gst/rtsp-server/rtsp-client.h:
9611 auth: add support for multiple basic auth tokens
9612 Make it possible to add multiple basic authorisation tokens to one authorization
9613 object. Associate with each token an authorization group that will define what
9614 capabilities are allowed.
9616 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9618 * gst/rtsp-server/rtsp-client.c:
9619 client: error out on non-aggregate control
9620 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9622 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9624 * gst/rtsp-server/rtsp-client.c:
9625 client: rework setup request a little
9626 Cache the media in DESCRIBE based on the longest matching path with the uri
9627 that we can find in the mount points.
9628 Rework the setup request a little to get the media from the session or from
9629 the longest matching path, this way we can derive the control string as
9630 everything after the path instead of hardcoding it.
9631 Find the stream based on the control string and only open a session when all
9634 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9636 * gst/rtsp-server/rtsp-media.c:
9637 * gst/rtsp-server/rtsp-media.h:
9638 media: add method to find a stream by control url
9640 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9642 * gst/rtsp-server/rtsp-stream.c:
9643 * gst/rtsp-server/rtsp-stream.h:
9644 stream: add method to check control url of stream
9646 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9648 * gst/rtsp-server/rtsp-client.c:
9649 * gst/rtsp-server/rtsp-session-media.c:
9650 * gst/rtsp-server/rtsp-session-media.h:
9651 * gst/rtsp-server/rtsp-session.c:
9652 * gst/rtsp-server/rtsp-session.h:
9653 session: use path matching for session media
9654 Use a path string instead of a uri to lookup session media in the sessions. Also
9655 use path matching to find the largest possible path that matches.
9657 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9659 * gst/rtsp-server/rtsp-client.c:
9660 * gst/rtsp-server/rtsp-mount-points.c:
9661 * gst/rtsp-server/rtsp-mount-points.h:
9662 * tests/check/gst/mountpoints.c:
9663 mount-points: remove useless vmethod
9664 Making lookups in the mount points should not be done with a URL, if there is a
9665 mapping to be done from URL to mount points, we'll need to do it somewhere
9668 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9670 * gst/rtsp-server/rtsp-mount-points.c:
9671 * gst/rtsp-server/rtsp-mount-points.h:
9672 * tests/check/gst/mountpoints.c:
9673 mount-points: improve mount point searching
9674 Use a GSequence to keep track of the mount points.
9675 Match a URL to the longest matching registered mount point. This should be the
9676 URL to perform aggreagate control and the remainder is the stream specific
9678 Add some unit tests for this.
9680 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9682 * gst/rtsp-server/Makefile.am:
9683 rtsp-server: Allow building of static library
9685 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9687 * tests/check/gst/mediafactory.c:
9688 tests: fix compilation
9690 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9692 * gst/rtsp-server/rtsp-sdp.c:
9693 sdp: get control string from stream
9694 Use the control string as configured in the stream.
9696 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9698 * gst/rtsp-server/rtsp-stream.c:
9699 * gst/rtsp-server/rtsp-stream.h:
9700 stream: add methods and property to set control string
9702 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9704 * gst/rtsp-server/rtsp-client.c:
9706 Rename variables for clarity
9707 Keep media in state when we can
9709 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9711 * gst/rtsp-server/rtsp-client.c:
9712 * gst/rtsp-server/rtsp-stream.c:
9713 * gst/rtsp-server/rtsp-stream.h:
9714 stream: add more support for IPv6
9715 Rename _get_address to _get_multicast_address in GstRTSPStream to
9716 make it clear that this function only deals with multicast.
9717 Make it possible to have both an IPv4 and IPv6 multicast address on
9718 a stream. Give the client an IPv4 or IPv6 address depending on the
9719 address it used to connect to the server.
9720 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9722 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9724 * gst/rtsp-server/rtsp-client.c:
9727 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9729 * gst/rtsp-server/rtsp-stream.c:
9730 stream: handle failed port allocation
9731 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9732 can't allocate any family at all. Also keep track of what port families we
9734 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9736 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9738 * gst/rtsp-server/rtsp-stream.c:
9739 stream: improve docs
9741 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9743 * gst/rtsp-server/rtsp-stream-transport.c:
9744 stream-transport: remove old if 0 block
9746 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9748 * tests/check/gst/client.c:
9750 gst_rtsp_client_get_uri() has been removed
9751 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9753 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9755 * gst/rtsp-server/rtsp-client.c:
9756 * gst/rtsp-server/rtsp-client.h:
9757 client: add method to filter managed sessions
9758 Add a method to filter the sessions managed by this client connection.
9759 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9761 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9763 * gst/rtsp-server/rtsp-client.c:
9764 * gst/rtsp-server/rtsp-client.h:
9765 client: remove _get_uri() method
9766 Remove the get_uri() method on the client. A client has no uri, the uri
9767 property is an internal property to manage the last cached media for
9770 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9772 * gst/rtsp-server/rtsp-media-factory.h:
9773 media-factory: fix typo
9775 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9777 * gst/rtsp-server/rtsp-media.c:
9778 rtsp-media: Do not leak the query in default_query_stop
9779 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9781 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9783 * gst/rtsp-server/rtsp-media.c:
9784 media: don't unlock when conversion fails
9785 Don't unlock the state lock when conversion fails because it was not locked.
9787 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9789 * gst/rtsp-server/rtsp-media.c:
9790 * gst/rtsp-server/rtsp-media.h:
9791 Add query_position and query_stop vmethods to rtsp-media
9793 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9795 * gst/rtsp-server/rtsp-media.c:
9796 Fix typo in property install for rtsp-media's time-provider
9798 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9800 * gst/rtsp-server/rtsp-client.c:
9801 * gst/rtsp-server/rtsp-client.h:
9802 client: clean some variables
9803 Clean some variables and add some guards to _send_request()
9805 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9807 * gst/rtsp-server/rtsp-client.c:
9808 * gst/rtsp-server/rtsp-client.h:
9809 Add gst_rtsp_client_send_request API
9810 This makes it possible to send arbitrary messages to a client, such as
9811 SET_PARAMETER or GET_PARAMETER
9813 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9815 * gst/rtsp-server/rtsp-media.c:
9816 * gst/rtsp-server/rtsp-media.h:
9817 media: add _get_element() method
9818 Add method to get the element used when creating the media.
9819 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9821 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9823 * gst/rtsp-server/rtsp-media.c:
9826 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9828 * gst/rtsp-server/rtsp-stream.c:
9829 * gst/rtsp-server/rtsp-stream.h:
9830 stream: allow access to the rtp session
9831 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9833 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9835 * gst/rtsp-server/rtsp-stream.c:
9836 * gst/rtsp-server/rtsp-stream.h:
9837 dscp qos support in gst-rtsp-stream
9838 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9840 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9842 * tests/check/gst/rtspserver.c:
9844 Actually do what the comment says. Also keep the old code around, not sure what
9845 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9846 it currently doesn't.
9848 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9850 * gst/rtsp-server/rtsp-client.c:
9851 client: also watch newly created session
9852 When we newly created a session, start watching it immediately instead of
9853 on the next request.
9855 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9857 * tests/check/gst/client.c:
9858 tests: add unit test for new-session
9859 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9861 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9863 * gst/rtsp-server/rtsp-client.c:
9864 client: emit new-session when new session is created
9865 Only emit new-session when we created a new session for a client, not when a
9866 client picked up a previous session.
9867 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9869 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9871 * gst/rtsp-server/rtsp-client.c:
9872 client: handle asterisk as path in requests
9873 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9875 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9877 * gst/rtsp-server/rtsp-media.c:
9878 media: handle segment query format mismatch
9879 It's possible that the segment query returns with a different format than what
9880 we asked for, handle this case also.
9882 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9884 * gst/rtsp-server/rtsp-media.c:
9885 media: use segment stop in collect_media_stats
9886 Use segment stop instead of duration as range end point.
9887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9889 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9891 * gst/rtsp-server/rtsp-media.c:
9892 * tests/check/gst/media.c:
9893 rtsp-media: Do not leak the element in take_pipeline
9894 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9896 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9898 * gst/rtsp-server/rtsp-client.c:
9899 * gst/rtsp-server/rtsp-client.h:
9900 rtsp-client: Make configure_client_transport virtual
9901 This patch makes configure_client_transport virtual. The functionality is
9902 needed to handle some weird clients sending multicast transport settings as url
9904 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9906 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9908 * gst/rtsp-server/rtsp-client.c:
9909 * gst/rtsp-server/rtsp-client.h:
9910 rtsp-client: Make param_set and param_get virtual
9911 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9913 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9915 * gst/rtsp-server/rtsp-client.c:
9916 * gst/rtsp-server/rtsp-media.c:
9917 * gst/rtsp-server/rtsp-media.h:
9918 media: convert_range replaces get_range_times
9919 get_range_times worked for handling UTC ranges for seeks, but we also
9920 need to convert back from NPT to the requested unit in
9921 get_range_string. convert_range is now used for both.
9922 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9924 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9926 * gst/rtsp-server/rtsp-client.c:
9927 * gst/rtsp-server/rtsp-sdp.c:
9928 * gst/rtsp-server/rtsp-sdp.h:
9929 sdp: cleanup sdp info
9930 We don't need to pass the proto, we can more easily check a boolean.
9931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9933 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9935 * gst/rtsp-server/rtsp-sdp.c:
9936 use 0.0.0.0 or :: for c= line instead of server address
9938 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9940 * gst/rtsp-server/rtsp-client.c:
9941 use local address, not remote, in SDP
9942 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9944 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9947 Automatic update of common submodule
9948 From 098c0d7 to 01a7a46
9950 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9952 * gst/rtsp-server/rtsp-media.c:
9953 * gst/rtsp-server/rtsp-media.h:
9954 media: possibility to override range time conversion
9955 Make it possible to override the conversion from GstRTSPTimeRange to
9956 GstClockTimes, that is done before seeking on the media
9957 pipeline. Overriding can be useful for UTC ranges, where the default
9958 conversion gives nanoseconds since 1900.
9959 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9961 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9963 * gst/rtsp-server/rtsp-server.c:
9964 * gst/rtsp-server/rtsp-server.h:
9965 rtsp-server: Expose the use_client_settings API
9966 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9968 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9970 * gst/rtsp-server/rtsp-client.c:
9971 * gst/rtsp-server/rtsp-stream.c:
9972 * gst/rtsp-server/rtsp-stream.h:
9973 rtspstream: handle both ipv4 and ipv6 clients
9974 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9976 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9978 * gst/rtsp-server/rtsp-sdp.c:
9979 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9980 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9981 We already have a way to place extra attributes in the SDP by using a string
9982 property with prefix x- or a- in the caps.
9984 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9986 * gst/rtsp-server/rtsp-sdp.c:
9987 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9988 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9989 We already have a way to place extra attributes in the SDP, just make a string
9990 property in the payloader with a- or x- prefix.
9992 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9994 * gst/rtsp-server/rtsp-sdp.c:
9995 rtsp: place a- and x- properties as attributes
9996 application/x-rtp has properties with a- and x- prefixes that should be
9997 placed as attributes in the SDP for the media instead of being added to the
10000 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10002 * examples/Makefile.am:
10003 * examples/test-video.c:
10004 example: add TLS example
10006 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10008 * gst/rtsp-server/rtsp-server.c:
10009 * gst/rtsp-server/rtsp-server.h:
10010 server: add support for TLS
10011 Add methods to set and get a TLS certificate.
10012 Add vmethod to configure a new connection. By default, configure the TLS
10013 certificate in a new connection if needed.
10015 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10017 * gst/rtsp-server/rtsp-server.c:
10018 * gst/rtsp-server/rtsp-server.h:
10019 server: remove accept_client vmethod
10020 This vmethod is not very useful so remove it.
10022 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10024 * gst/rtsp-server/rtsp-server.c:
10025 server: don't crash on NULL GError
10027 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
10029 * gst/rtsp-server/rtsp-session-pool.c:
10030 rtsp-session-pool: corrected session timeout detection
10031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
10033 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10035 * gst/rtsp-server/rtsp-client.c:
10036 client: improve debug
10038 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10040 * gst/rtsp-server/rtsp-client.c:
10041 * gst/rtsp-server/rtsp-client.h:
10042 * gst/rtsp-server/rtsp-server.c:
10043 server: refactor connection setup
10044 Let the server accept the socket connection and construct a GstRTSPConnection
10045 from it. Remove the code from the client and let the client only deal with
10046 a fully configure GstRTSPConnection object.
10047 We will need this later when the server will configure the connection for
10050 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10052 * gst/rtsp-server/rtsp-stream.c:
10053 stream: keep the transport object alive
10054 Keep the transport object alive while we have it as qdata on the
10057 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
10059 * gst/rtsp-server/rtsp-client.c:
10060 * gst/rtsp-server/rtsp-server.c:
10061 rtsp-server: Do not crash on nmapping of server
10062 * generate error when gst_rtsp_connection_accept fails
10063 * do not stop accepting incoming connections because
10064 accepting a client fails
10065 https://bugzilla.gnome.org/show_bug.cgi?id=701072
10067 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
10069 * gst/rtsp-server/rtsp-client.c:
10070 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
10071 https://bugzilla.gnome.org/show_bug.cgi?id=700953
10073 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
10075 * gst/rtsp-server/rtsp-sdp.c:
10076 rtsp-sdp: Parse framerate caps field and set SDP attribute
10077 The SDP attribute and its format is described in RFC4566.
10078 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10080 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
10082 * gst/rtsp-server/rtsp-sdp.c:
10083 rtsp-sdp: Parse width/height from caps and set SDP attribute
10084 The SDP attribute and its format is described in RFC6064.
10085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10087 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
10089 * gst/rtsp-server/rtsp-sdp.c:
10090 * tests/check/gst/client.c:
10091 rtsp-sdp: add bandwidth line
10092 https://bugzilla.gnome.org/show_bug.cgi?id=699220
10094 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10097 Automatic update of common submodule
10098 From 5edcd85 to 098c0d7
10100 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10102 * tests/check/gst/media.c:
10103 tests: add dynamic payloader prepare/unprepare check
10105 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10107 * gst/rtsp-server/rtsp-media.c:
10108 media: release lock when removing fakesink
10110 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10112 * gst/rtsp-server/rtsp-stream.c:
10113 stream: set elements to NULL before removing
10114 When removing a stream, set the elements to NULL first. This avoids
10115 element-is-not-in-NULL-state errors when we dispose the elements.
10117 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
10120 Automatic update of common submodule
10121 From 3cb3d3c to 5edcd85
10123 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10125 * gst/rtsp-server/rtsp-media.c:
10126 * gst/rtsp-server/rtsp-media.h:
10127 media: listen to pad-removed signals
10128 Listen to the pad-removed signal and remove the stream associated with the
10130 Add signal to be notified of the removed pad.
10131 Remove the fakesink in unprepare()
10132 Fix signatures of the signal methods
10134 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10136 * examples/test-sdp.c:
10137 tests: add example of reusable pipelines
10139 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
10141 * gst/rtsp-server/rtsp-stream.c:
10142 * gst/rtsp-server/rtsp-stream.h:
10143 stream: add method to get the srcpad
10145 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10147 * tests/check/gst/media.c:
10148 check: add media prepare/unprepare test
10149 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10151 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
10153 * gst/rtsp-server/rtsp-media.c:
10154 media: disconnect from signal handlers in unprepare()
10155 We connected to the pad-added and no-more-pads signals in prepare() so
10156 we need to disconnect from them in unprepare().
10157 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10159 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
10161 * gst/rtsp-server/rtsp-media.c:
10162 media: don't free streams array
10163 Don't free the streams array in the unprepare() method, they were not
10164 added in prepare().
10165 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10167 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
10169 * gst/rtsp-server/rtsp-media.c:
10170 media: don't unref the pipeline in unprepare
10171 Unprepare() should undo what prepare() does. Because the pipeline is
10172 not created in prepare(), we should not unref it in unprepare()
10174 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
10176 * gst/rtsp-server/rtsp-stream.c:
10177 stream: clear session and caps for reuse
10178 Set the session and caps to NULL after unref otherwise we might unref
10180 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10182 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
10184 * gst/rtsp-server/rtsp-client.c:
10185 client: send out teardown signal before tearing down
10186 The advantage is that in the signal handler you get direct access to
10187 information about what streams are about to get torn down (in the
10188 GstRTSPClientState).
10189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
10191 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
10193 * gst/rtsp-server/rtsp-client.c:
10194 * gst/rtsp-server/rtsp-client.h:
10195 client: expose connection
10196 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
10198 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
10201 Automatic update of common submodule
10202 From aed87ae to 3cb3d3c
10204 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10206 * gst/rtsp-server/rtsp-media.c:
10207 * gst/rtsp-server/rtsp-media.h:
10208 * gst/rtsp-server/rtsp-session-media.c:
10209 * gst/rtsp-server/rtsp-session-media.h:
10210 media: add method to get the base_time of the pipeline
10211 Together with a shared clock, this base-time could eventually be sent to
10212 the client so that it can reconstruct the exact running-time of the clock
10215 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10217 * gst/rtsp-server/Makefile.am:
10218 * gst/rtsp-server/rtsp-media.c:
10219 * gst/rtsp-server/rtsp-media.h:
10220 * gst/rtsp-server/rtsp-sdp.c:
10221 media: add GstNetTimeProvider support
10222 Add a property to let the media provide a GstNetTimeProvider for its clock.
10223 Make methods to get the clock and nettimeprovider
10224 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
10225 provider and also the current time of the clock. This should make it possible
10226 for (GStreamer) clients to slave their clock to the server clock.
10228 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
10231 Automatic update of common submodule
10232 From 04c7a1e to aed87ae
10234 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10236 * gst/rtsp-server/rtsp-media.c:
10237 media: wait for buffering to complete
10238 Wait for buffering to complete before changing the state to the target state.
10240 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10242 * gst/rtsp-server/rtsp-media.c:
10243 media: small cleanup
10245 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
10247 * tests/check/gst/rtspserver.c:
10248 tests: remove extra unref in test_setup_non_existing_stream
10249 The unref is not needed anymore, teardown runs without it.
10250 https://bugzilla.gnome.org/show_bug.cgi?id=696542
10252 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
10254 * tests/check/gst/rtspserver.c:
10255 tests: GSocketService cleanup in test_bind_already_in_use
10256 Use g_socket_service_stop so the rtspserver test stops listening for
10257 incoming connections in test_bind_already_in_use.
10258 https://bugzilla.gnome.org/show_bug.cgi?id=696541
10260 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
10262 * gst/rtsp-server/rtsp-media-factory.c:
10263 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
10264 Instead use a GWeakRef which is safe to use
10265 This is a known GLib bug, see:
10266 https://bugzilla.gnome.org/show_bug.cgi?id=667145
10268 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
10270 * gst/rtsp-server/rtsp-client.c:
10271 * gst/rtsp-server/rtsp-media.c:
10272 * gst/rtsp-server/rtsp-media.h:
10273 * gst/rtsp-server/rtsp-sdp.c:
10274 * tests/check/gst/media.c:
10275 * tests/check/gst/rtspserver.c:
10276 rtsp-media/client: Reply to PLAY request with same type of Range
10277 Remember the type of Range from the PLAY request and use the same type for
10280 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
10282 * gst/rtsp-server/rtsp-client.c:
10283 * gst/rtsp-server/rtsp-client.h:
10284 * tests/check/gst/client.c:
10285 rtsp-client: expose uri
10287 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
10289 * tests/check/gst/mediafactory.c:
10290 tests: Hold ref while creating second media
10291 To test if the media aren't shared, make sure we keep the first one while creating a second
10292 otherwise the same memory address may be reused.
10294 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
10297 configure: remove out-of-date comment
10299 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10302 .gitignore: ignore more build files
10304 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10306 * tests/check/Makefile.am:
10307 tests: use right _LIBS variable for gst-plugins-base libs
10309 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10311 * tests/check/Makefile.am:
10312 check: add librtp to libs
10314 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10316 * tests/check/gst/rtspserver.c:
10317 tests: Add test to check selecting a port the server will send from
10319 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10321 * tests/check/gst/rtspserver.c:
10322 tests: Make sure packets are actually received
10324 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10326 * gst/rtsp-server/rtsp-stream.c:
10327 stream: Select unicast address from pool if appropriate
10329 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10331 * gst/rtsp-server/rtsp-stream.c:
10332 stream: Properties are always there in Gst 1.0
10334 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10336 * tests/check/gst/addresspool.c:
10337 tests: Add tests for unicast addresses in pool
10339 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10341 * gst/rtsp-server/rtsp-address-pool.c:
10342 * tests/check/gst/addresspool.c:
10343 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10345 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10347 * docs/libs/gst-rtsp-server-sections.txt:
10348 * gst/rtsp-server/rtsp-address-pool.c:
10349 * gst/rtsp-server/rtsp-address-pool.h:
10350 * gst/rtsp-server/rtsp-stream.c:
10351 * tests/check/gst/addresspool.c:
10352 address-pool: Add unicast addresses
10354 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10357 * gst/rtsp-server/rtsp-server.c:
10358 * tests/check/gst/rtspserver.c:
10359 rtsp-server: Limit the number of threads per server instance
10360 If we exceed the maximum, just round robin the clients over the existing
10363 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10365 * gst/rtsp-server/rtsp-server.c:
10366 rtsp-server: No need to store the GMainContext in the client context
10368 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10370 * tests/check/gst/rtspserver.c:
10371 tests: Add test for client disconnection
10373 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10375 * tests/check/gst/rtspserver.c:
10376 tests: Test client and session timeouts with multiple threads
10378 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10380 * gst/rtsp-server/rtsp-address-pool.c:
10381 * gst/rtsp-server/rtsp-auth.c:
10382 * gst/rtsp-server/rtsp-client.c:
10383 * gst/rtsp-server/rtsp-media-factory-uri.c:
10384 * gst/rtsp-server/rtsp-media-factory.c:
10385 * gst/rtsp-server/rtsp-media.c:
10386 * gst/rtsp-server/rtsp-mount-points.c:
10387 * gst/rtsp-server/rtsp-server.c:
10388 * gst/rtsp-server/rtsp-session-media.c:
10389 * gst/rtsp-server/rtsp-session-pool.c:
10390 * gst/rtsp-server/rtsp-session.c:
10391 Document locking and its order
10393 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10395 * tests/check/gst/rtspserver.c:
10396 tests: Test that slow DESCRIBE don't block other clients
10398 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10400 * tests/check/gst/client.c:
10401 tests: Add tests for client-requested multicast address
10403 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10405 * docs/libs/gst-rtsp-server-sections.txt:
10406 docs: Put the various functions in the right sections
10408 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10410 * docs/libs/gst-rtsp-server-docs.sgml:
10411 * docs/libs/gst-rtsp-server-sections.txt:
10412 * gst/rtsp-server/rtsp-address-pool.c:
10413 * gst/rtsp-server/rtsp-address-pool.h:
10414 docs: Generate docs for GstRTSPAddressPool
10416 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10418 * gst/rtsp-server/rtsp-client.c:
10419 * gst/rtsp-server/rtsp-stream.c:
10420 * gst/rtsp-server/rtsp-stream.h:
10421 client: Check client provided addresses against the address pool
10423 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10425 * gst/rtsp-server/rtsp-address-pool.c:
10426 * gst/rtsp-server/rtsp-address-pool.h:
10427 * tests/check/gst/addresspool.c:
10428 address-pool: Add API to request a specific address from the pool
10429 Also add relevant unit tests.
10431 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10433 * tests/check/gst/mediafactory.c:
10434 tests: Check the passing around of a RTSPAddressPool
10435 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10436 way down to the stream.
10438 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10440 * tests/check/gst/addresspool.c:
10441 tests: Add more tests for the address pool
10443 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10445 * gst/rtsp-server/rtsp-address-pool.c:
10446 address-pool: Fix off by one error
10447 When splitting a port range, the port after a skip is not part of range.
10449 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10452 Automatic update of common submodule
10453 From 2de221c to 04c7a1e
10455 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10458 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10459 AM_CONFIG_HEADER was removed in automake 1.13
10460 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10462 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10465 Automatic update of common submodule
10466 From a942293 to 2de221c
10468 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10470 * gst/rtsp-server/rtsp-client.c:
10471 client: make sure the watch exists while sending data
10472 Protect the send_func with a lock. This allows us to wait for sending
10473 to complete before changing the send_func and user_data. We add an
10474 extra ref to the watch to make sure that it remains valid during
10476 When closing the connection, set the send_func to NULL
10477 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10479 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10481 * tests/check/Makefile.am:
10482 tests: use GST_*_1_0 environment variables everywhere
10483 The _1_0 suffixed environment variables override the
10484 non-suffixed ones, so if we're in an environment that
10485 sets the _1_0 suffixed ones, such as jhbuild, we need
10486 to set those to make sure ours actually always get
10489 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10492 Automatic update of common submodule
10493 From acb04d9 to a942293
10495 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10497 * gst/rtsp-server/rtsp-client.c:
10498 rtsp-client: set the client backlog
10499 Set the client backlog to a reasonable default
10501 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10503 * gst/rtsp-server/rtsp-media.c:
10504 rtsp-media: Make the element a constructor parameter
10505 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10507 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10509 * docs/libs/Makefile.am:
10510 docs: Link with gcov library when gcov is enabled
10511 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10513 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10515 * gst/rtsp-server/rtsp-media.c:
10516 media: match prepare with unprepare
10517 Really unprepare when there were an equal amount of prepare calls.
10519 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10521 * gst/rtsp-server/rtsp-media.c:
10522 media: media has to be unprepared in finalize
10523 Because unprepare takes away the last ref on the media.
10525 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10527 * gst/rtsp-server/rtsp-client.c:
10528 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10529 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10530 We can't use the refcount to trigger unprepare because it is the unprepare call
10531 that removes the last refcount after all messages are consumed. What we should
10532 probably do is make a prepared refcount and only unprepare when the refcount
10535 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10537 * gst/rtsp-server/rtsp-media.c:
10538 media: let the source unref the last media ref
10539 the last ref to the media is held by the source so we don't need to add more ref
10540 and unrefs, we simply destroy the media when the source is gone.
10542 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10544 * gst/rtsp-server/rtsp-media.c:
10545 media: improve debug
10547 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10549 * gst/rtsp-server/rtsp-media.c:
10551 Make sure we are in the right state when collecting the position and duration.
10552 Only make ourselves PREPARED when we were previously PREPARING.
10554 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10556 * gst/rtsp-server/rtsp-media.c:
10557 media: use g_object_ref/unref for GObjects
10559 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10561 * gst/rtsp-server/rtsp-client.c:
10562 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10563 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10564 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10565 isn't being used anymore.
10567 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10569 * gst/rtsp-server/rtsp-media.c:
10570 Fix compiler warning
10572 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10574 * gst/rtsp-server/rtsp-media-factory-uri.c:
10575 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10577 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10579 * gst/rtsp-server/rtsp-session-media.h:
10582 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10584 * gst/rtsp-server/rtsp-media.c:
10585 * tests/check/gst/media.c:
10586 media: avoid element leak
10588 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10590 * gst/rtsp-server/rtsp-media.c:
10591 media: require an element in media constructor
10593 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10595 * gst/rtsp-server/rtsp-client.c:
10596 Revert "client: TEARDOWN brings that state to Init again"
10597 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10598 The object is already disposed, there is no point in setting the state.
10600 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10602 * gst/rtsp-server/rtsp-client.c:
10603 client: TEARDOWN brings that state to Init again
10605 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * docs/libs/gst-rtsp-server-sections.txt:
10608 * examples/test-auth.c:
10609 * gst/rtsp-server/rtsp-auth.c:
10610 * gst/rtsp-server/rtsp-auth.h:
10611 * gst/rtsp-server/rtsp-client.c:
10612 * gst/rtsp-server/rtsp-client.h:
10613 * gst/rtsp-server/rtsp-media-factory-uri.c:
10614 * gst/rtsp-server/rtsp-media-factory-uri.h:
10615 * gst/rtsp-server/rtsp-media-factory.c:
10616 * gst/rtsp-server/rtsp-media-factory.h:
10617 * gst/rtsp-server/rtsp-media.c:
10618 * gst/rtsp-server/rtsp-media.h:
10619 * gst/rtsp-server/rtsp-mount-points.c:
10620 * gst/rtsp-server/rtsp-mount-points.h:
10621 * gst/rtsp-server/rtsp-sdp.c:
10622 * gst/rtsp-server/rtsp-server.c:
10623 * gst/rtsp-server/rtsp-server.h:
10624 * gst/rtsp-server/rtsp-session-media.c:
10625 * gst/rtsp-server/rtsp-session-media.h:
10626 * gst/rtsp-server/rtsp-session-pool.c:
10627 * gst/rtsp-server/rtsp-session-pool.h:
10628 * gst/rtsp-server/rtsp-session.c:
10629 * gst/rtsp-server/rtsp-session.h:
10630 * gst/rtsp-server/rtsp-stream-transport.c:
10631 * gst/rtsp-server/rtsp-stream-transport.h:
10632 * gst/rtsp-server/rtsp-stream.c:
10633 * gst/rtsp-server/rtsp-stream.h:
10634 * tests/check/gst/media.c:
10635 rtsp: make object details private
10636 Make all object details private
10637 Add methods to access private bits
10639 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10641 * tests/check/Makefile.am:
10642 * tests/check/gst/media.c:
10643 tests: add media tests
10645 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10647 * gst/rtsp-server/rtsp-media.c:
10648 media: check if prepared for some methods
10649 Check that the media object is prepared before doing seek and getting the
10650 current position etc.
10651 Add some g_return checks.
10653 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10655 * tests/check/Makefile.am:
10656 * tests/check/gst/mediafactory.c:
10657 tests: add mediafactory test
10659 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10661 * gst/rtsp-server/rtsp-stream.c:
10662 stream: improve debug
10664 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10666 * gst/rtsp-server/rtsp-media.c:
10667 * gst/rtsp-server/rtsp-media.h:
10668 media: unref pipeline in finalize to avoid leaking it
10670 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10672 * gst/rtsp-server/rtsp-media-factory-uri.c:
10673 * gst/rtsp-server/rtsp-media.c:
10674 rtsp: use gst_object_unref on GstObjects
10676 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10678 * gst/rtsp-server/rtsp-media-factory.c:
10679 media-factory: require an url
10681 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10683 * examples/test-uri.c:
10684 examples: fix include
10686 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10688 * gst/rtsp-server/rtsp-server.h:
10689 server: remove unused include
10691 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10693 * tests/check/Makefile.am:
10694 * tests/check/gst/mountpoints.c:
10695 tests: add test for mountpoints
10697 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10699 * gst/rtsp-server/rtsp-client.c:
10700 client: fix factory leak
10701 Keep the factory in the state object only for authorization checks and make
10702 sure we unref it on failure. Also don't keep invalid objects in the state
10705 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10707 * gst/rtsp-server/rtsp-mount-points.c:
10708 mounts: add g_return_if guards
10710 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10712 * tests/check/gst/client.c:
10713 tests: add more tests
10715 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10717 * gst/rtsp-server/rtsp-client.c:
10718 client: improve debug
10720 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10722 * gst/rtsp-server/rtsp-client.c:
10723 client: improve debug and fix leaks
10724 Cleanup the uri and session when there is a bad request.
10726 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10731 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10733 * tests/check/gst/client.c:
10734 test: add test for session in options request
10736 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10738 * gst/rtsp-server/rtsp-client.c:
10739 client: use 454 when session can't be found
10740 We should use 454 when a session can't be found because there was no session
10741 pool configured in the server. This is not a server configuration problem
10742 because the server on which the request is done might not be the same one that
10743 will keep the sessions for us and so it does not need to support sessions.
10745 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10747 * gst/rtsp-server/rtsp-client.c:
10748 client: only free connection when there is one
10749 It's possible that the client doesn't have a connection when we try to free it.
10751 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10753 * tests/check/Makefile.am:
10754 * tests/check/gst/client.c:
10755 tests: add unit test for the client object
10757 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10759 * gst/rtsp-server/rtsp-client.c:
10760 client: small cleanup
10762 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10764 * gst/rtsp-server/rtsp-client.h:
10765 client: remove unused include
10767 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10769 * gst/rtsp-server/rtsp-client.c:
10770 client: fix compilation
10772 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10774 * gst/rtsp-server/rtsp-client.c:
10775 client: call destroy without the lock
10777 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10779 * gst/rtsp-server/rtsp-client.c:
10780 * gst/rtsp-server/rtsp-client.h:
10781 client: make the client usable without a socket
10782 Make a method to let the client handle a message and a callback when the client
10783 wants us to send a response message back. This makes it possible to also use the
10784 client object without the sockets, which should make it easier to test.
10786 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10788 * gst/rtsp-server/rtsp-client.c:
10789 * gst/rtsp-server/rtsp-client.h:
10790 client: small cleanup
10792 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10794 * docs/libs/gst-rtsp-server-sections.txt:
10795 * gst/rtsp-server/rtsp-client.c:
10796 * gst/rtsp-server/rtsp-client.h:
10797 * gst/rtsp-server/rtsp-server.c:
10798 client: remove reference to server
10799 We don't need to keep a ref to the server
10801 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10803 * gst/rtsp-server/rtsp-client.c:
10804 * gst/rtsp-server/rtsp-client.h:
10805 client: add locking
10806 Also add some g_return_if()
10808 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10810 * gst/rtsp-server/rtsp-client.c:
10811 client: log more errors
10813 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10815 * gst/rtsp-server/rtsp-client.c:
10816 client: fix compilation
10818 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10820 * gst/rtsp-server/rtsp-client.c:
10821 * gst/rtsp-server/rtsp-client.h:
10822 client: add generic close-after-send support
10823 Add a property to send_response() to close the connection after the response has
10824 been sent to the client.
10826 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10829 * docs/libs/gst-rtsp-server-docs.sgml:
10830 * docs/libs/gst-rtsp-server-sections.txt:
10831 * docs/libs/gst-rtsp-server.types:
10832 * examples/test-auth.c:
10833 * examples/test-launch.c:
10834 * examples/test-mp4.c:
10835 * examples/test-multicast.c:
10836 * examples/test-multicast2.c:
10837 * examples/test-ogg.c:
10838 * examples/test-readme.c:
10839 * examples/test-sdp.c:
10840 * examples/test-uri.c:
10841 * examples/test-video.c:
10842 * gst/rtsp-server/Makefile.am:
10843 * gst/rtsp-server/rtsp-auth.h:
10844 * gst/rtsp-server/rtsp-client.c:
10845 * gst/rtsp-server/rtsp-client.h:
10846 * gst/rtsp-server/rtsp-media-mapping.c:
10847 * gst/rtsp-server/rtsp-media-mapping.h:
10848 * gst/rtsp-server/rtsp-mount-points.c:
10849 * gst/rtsp-server/rtsp-mount-points.h:
10850 * gst/rtsp-server/rtsp-server.c:
10851 * gst/rtsp-server/rtsp-server.h:
10852 * gst/rtsp-server/rtsp-session-media.c:
10853 * gst/rtsp-server/rtsp-session-pool.c:
10854 * gst/rtsp-server/rtsp-session-pool.h:
10855 * tests/check/gst/rtspserver.c:
10856 MediaMapping -> MountPoints
10857 Describes better what the object manages.
10859 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10862 configure: bump required version of -base
10864 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10866 * gst/rtsp-server/rtsp-media.c:
10869 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10871 * gst/rtsp-server/rtsp-media.c:
10872 * gst/rtsp-server/rtsp-media.h:
10873 media: support more Range formats
10874 Use the new -base methods to convert the Range string into a seek start and stop
10877 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10879 * examples/test-launch.c:
10880 examples: fix whitespace
10882 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10884 * examples/test-auth.c:
10885 test-auth: add example of how to remove sessions
10886 Add an example of the session filter api.
10888 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10890 * examples/test-uri.c:
10891 test-uri: remove mapping example
10893 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10895 * examples/test-uri.c:
10896 test-uri: fix callback signature
10898 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10900 * gst/rtsp-server/rtsp-media-factory.c:
10901 factory: keep ref to factory while media active
10902 While the media from a factory is alive, keep a ref to the factory.
10903 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10905 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10907 * gst/rtsp-server/rtsp-media-factory-uri.c:
10908 factory-uri: add some debug
10910 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10912 * gst/rtsp-server/rtsp-stream.c:
10913 stream: set udp sources to PLAYING
10914 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10915 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10917 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10919 * gst/rtsp-server/rtsp-media-factory-uri.c:
10920 factory-uri: take ref to factory
10921 Take a ref to the factory that we place in our list.
10923 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10925 * tests/Makefile.am:
10926 * tests/test-reuse.c:
10927 test: add test for server reuse
10928 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10930 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10932 * gst/rtsp-server/rtsp-server.c:
10933 server: start and stop multiple times
10934 Stop listening on the RTSP port when the GSource is removed, so clients
10935 can't connect and the server can be started again.
10936 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10938 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10940 * gst/rtsp-server/rtsp-server.c:
10941 server: fix small leak
10943 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10945 * gst/rtsp-server/rtsp-media.c:
10946 media: unref source in finish_unprepare
10947 The source is created in prepare, unref it in finish_unprepare.
10948 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10950 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10952 * gst/rtsp-server/rtsp-client.c:
10953 * gst/rtsp-server/rtsp-media.c:
10954 rtsp-media: remove bus watch before finalizing
10955 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10956 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10957 the GDestroyNotify function.
10958 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10959 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10960 gst_rtsp_media_unprepare before unreffing the media.
10961 This way, the bus watch will be removed before the media is finalized.
10962 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10964 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10966 * gst/rtsp-server/rtsp-client.c:
10967 * gst/rtsp-server/rtsp-client.h:
10968 client: wait until the TEARDOWN response is sent to close the connection
10969 Responses can be sent async so we need to wait until the TEARDOWN response has
10970 been written before we close the connection to the client. This avoids the risk
10971 of writing/polling closed sockets.
10972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10974 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10976 * gst/rtsp-server/rtsp-stream.c:
10977 rtsp-stream: plug socket leak
10978 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10980 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10983 Automatic update of common submodule
10984 From 6bb6951 to a72faea
10986 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10988 * gst/rtsp-server/rtsp-media-factory-uri.c:
10989 rtsp-server: don't use deprecated API
10991 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10993 * gst/rtsp-server/rtsp-client.c:
10994 rtsp-client: fix unused-but-set-variable compiler warning
10995 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10997 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11000 * docs/libs/gst-rtsp-server-sections.txt:
11001 * gst/rtsp-server/rtsp-client.c:
11004 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11006 * examples/Makefile.am:
11007 * examples/test-multicast2.c:
11008 examples: add another multicast example
11009 Add an example for how to configure separate multicast ranges for each media
11012 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11014 * examples/test-multicast.c:
11017 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11019 * gst/rtsp-server/rtsp-client.c:
11020 * gst/rtsp-server/rtsp-media.c:
11021 * gst/rtsp-server/rtsp-session-media.c:
11022 * gst/rtsp-server/rtsp-session-media.h:
11023 * gst/rtsp-server/rtsp-stream-transport.c:
11024 * gst/rtsp-server/rtsp-stream-transport.h:
11025 stream: use the address managed by the stream
11026 Use the address managed by the stream for multicast. This allows us to have 1
11027 multicast address for each stream.
11028 Because the address is now managed by the stream we don't have to pass it around
11030 Set the address pool on the streams.
11032 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11034 * gst/rtsp-server/rtsp-client.c:
11035 * gst/rtsp-server/rtsp-media.c:
11036 * gst/rtsp-server/rtsp-stream.c:
11037 rtsp: improve debug
11039 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11041 * gst/rtsp-server/rtsp-media.c:
11042 * gst/rtsp-server/rtsp-media.h:
11043 media: add signal for new streams
11044 This allows applications to listen for new streams and configure properties on
11045 them, like the address pool.
11047 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11049 * gst/rtsp-server/rtsp-media.c:
11050 media: configure address pool in new streams
11052 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11054 * gst/rtsp-server/rtsp-stream.c:
11055 * gst/rtsp-server/rtsp-stream.h:
11056 stream: add methods to deal with address pool
11057 Add methods to get and set the address pool for the stream
11058 Add method to allocate and get the multicast addresses for this stream.
11060 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11062 * docs/libs/gst-rtsp-server-sections.txt:
11063 * gst/rtsp-server/rtsp-media.c:
11064 * gst/rtsp-server/rtsp-media.h:
11065 media: remove MTU property
11066 It is a stream property
11068 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11070 * gst/rtsp-server/rtsp-client.c:
11071 client: set blocksize only on stream
11072 Set the blocksize only on the current stream.
11074 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11076 * gst/rtsp-server/rtsp-stream.c:
11077 stream: share src and sink sockets
11078 the allocated socket is in the used-socket property, not socket.
11080 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11082 * gst/rtsp-server/rtsp-address-pool.c:
11083 * gst/rtsp-server/rtsp-address-pool.h:
11084 * gst/rtsp-server/rtsp-client.c:
11085 * gst/rtsp-server/rtsp-session-media.c:
11086 * gst/rtsp-server/rtsp-session-media.h:
11087 * gst/rtsp-server/rtsp-stream-transport.c:
11088 * gst/rtsp-server/rtsp-stream-transport.h:
11089 * tests/check/gst/addresspool.c:
11090 rtsp: make address-pool return an address object
11091 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
11092 store more info in the structure and allows us to more easily return the address
11093 to the right pool when no longer needed.
11094 Pass the address to the StreamTransport so that we can return it to the pool
11095 when the stream transport is freed or changed.
11097 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11099 * examples/Makefile.am:
11100 * examples/test-multicast.c:
11101 examples: add multicast example
11102 Show how to set up the multicast address pool so that media can be
11103 server with multicast.
11105 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11107 * gst/rtsp-server/rtsp-client.c:
11108 * gst/rtsp-server/rtsp-media-factory.c:
11109 * gst/rtsp-server/rtsp-media-factory.h:
11110 * gst/rtsp-server/rtsp-media.c:
11111 * gst/rtsp-server/rtsp-media.h:
11112 rtsp: use AddressPool
11113 Remove the multicast_group property.
11114 Use the configured addresspool to allocate multicast addresses.
11116 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11118 * gst/rtsp-server/rtsp-address-pool.c:
11119 * gst/rtsp-server/rtsp-address-pool.h:
11120 address-pool: add clear method
11122 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11124 * gst/rtsp-server/rtsp-address-pool.c:
11125 address-pool: small cleanups
11127 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11129 * tests/check/Makefile.am:
11130 * tests/check/gst/addresspool.c:
11131 tests: add addresspool unit test
11133 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11135 * gst/rtsp-server/Makefile.am:
11136 * gst/rtsp-server/rtsp-address-pool.c:
11137 * gst/rtsp-server/rtsp-address-pool.h:
11138 address-pool: add object to manage multicast addresses
11139 Make an object that can manage a rage of multicast addresses and ports.
11141 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11143 * gst/rtsp-server/rtsp-server.c:
11144 server: set default max-threads property
11146 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11148 * gst/rtsp-server/rtsp-media.c:
11149 media: wait for concurrent _prepare
11150 If a prepare is busy, wait for the result.
11152 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11154 * gst/rtsp-server/rtsp-media.c:
11155 media: add lock around message handler
11156 We don't want to dispatch messages while we are still processing the result of
11159 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11161 * gst/rtsp-server/rtsp-media.c:
11162 * gst/rtsp-server/rtsp-media.h:
11163 media: add lock to protect state changes
11165 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11167 * gst/rtsp-server/rtsp-stream.c:
11168 * gst/rtsp-server/rtsp-stream.h:
11169 stream: add locking
11171 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11173 * gst/rtsp-server/rtsp-stream-transport.c:
11174 * gst/rtsp-server/rtsp-stream-transport.h:
11175 * gst/rtsp-server/rtsp-stream.c:
11176 stream-transport: add keep-alive method
11178 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11180 * gst/rtsp-server/rtsp-stream-transport.c:
11181 * gst/rtsp-server/rtsp-stream-transport.h:
11182 * gst/rtsp-server/rtsp-stream.c:
11183 stream-transport: add method to handle RTP/RTCP
11184 Call new methods instead of poking into the structures directly.
11186 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11188 * gst/rtsp-server/rtsp-session-media.c:
11189 * gst/rtsp-server/rtsp-session-media.h:
11190 session-media: add locking
11192 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11194 * gst/rtsp-server/rtsp-session.c:
11195 * gst/rtsp-server/rtsp-session.h:
11196 session: add locking
11198 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11200 * gst/rtsp-server/rtsp-server.c:
11201 server: free old socket
11203 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11205 * gst/rtsp-server/rtsp-media-mapping.c:
11206 * gst/rtsp-server/rtsp-media-mapping.h:
11207 mapping: add locking
11209 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11211 * gst/rtsp-server/rtsp-media-factory.c:
11212 media-factory: add locking
11214 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11216 * gst/rtsp-server/rtsp-auth.c:
11217 * gst/rtsp-server/rtsp-auth.h:
11220 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11222 * gst/rtsp-server/rtsp-server.c:
11223 * gst/rtsp-server/rtsp-server.h:
11224 server: add max-thread property
11226 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11228 * gst/rtsp-server/rtsp-server.c:
11229 * gst/rtsp-server/rtsp-server.h:
11230 server: use a threadpool for the mainloops
11232 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11234 * gst/rtsp-server/rtsp-client.c:
11235 * gst/rtsp-server/rtsp-client.h:
11236 client: rename method
11237 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
11238 don't really create the client from the socket, we use the socket for the
11241 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11243 * gst/rtsp-server/rtsp-client.c:
11244 * gst/rtsp-server/rtsp-client.h:
11245 * gst/rtsp-server/rtsp-server.c:
11246 server: rework maincontext handling in clients
11247 Make a separate method to attach a client to a MainContext.
11248 Let the server decide in what GMainContext the client will operate and give this
11249 context to the client in attach. Then the server can later decide to use a
11250 separate thread for each client or just use the mainthread.
11252 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11254 * gst/rtsp-server/rtsp-client.c:
11255 * gst/rtsp-server/rtsp-session.c:
11256 * gst/rtsp-server/rtsp-session.h:
11257 session: move session header code in session object
11259 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
11263 * examples/test-auth.c:
11264 * examples/test-launch.c:
11265 * examples/test-mp4.c:
11266 * examples/test-ogg.c:
11267 * examples/test-readme.c:
11268 * examples/test-sdp.c:
11269 * examples/test-uri.c:
11270 * examples/test-video.c:
11271 * gst/rtsp-server/rtsp-auth.c:
11272 * gst/rtsp-server/rtsp-auth.h:
11273 * gst/rtsp-server/rtsp-client.c:
11274 * gst/rtsp-server/rtsp-client.h:
11275 * gst/rtsp-server/rtsp-media-factory-uri.c:
11276 * gst/rtsp-server/rtsp-media-factory-uri.h:
11277 * gst/rtsp-server/rtsp-media-factory.c:
11278 * gst/rtsp-server/rtsp-media-factory.h:
11279 * gst/rtsp-server/rtsp-media-mapping.c:
11280 * gst/rtsp-server/rtsp-media-mapping.h:
11281 * gst/rtsp-server/rtsp-media.c:
11282 * gst/rtsp-server/rtsp-media.h:
11283 * gst/rtsp-server/rtsp-params.c:
11284 * gst/rtsp-server/rtsp-params.h:
11285 * gst/rtsp-server/rtsp-sdp.c:
11286 * gst/rtsp-server/rtsp-sdp.h:
11287 * gst/rtsp-server/rtsp-server.c:
11288 * gst/rtsp-server/rtsp-server.h:
11289 * gst/rtsp-server/rtsp-session-media.c:
11290 * gst/rtsp-server/rtsp-session-media.h:
11291 * gst/rtsp-server/rtsp-session-pool.c:
11292 * gst/rtsp-server/rtsp-session-pool.h:
11293 * gst/rtsp-server/rtsp-session.c:
11294 * gst/rtsp-server/rtsp-session.h:
11295 * gst/rtsp-server/rtsp-stream-transport.c:
11296 * gst/rtsp-server/rtsp-stream-transport.h:
11297 * gst/rtsp-server/rtsp-stream.c:
11298 * gst/rtsp-server/rtsp-stream.h:
11299 * tests/check/gst/rtspserver.c:
11300 * tests/test-cleanup.c:
11303 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11305 * gst/rtsp-server/rtsp-media.c:
11306 * gst/rtsp-server/rtsp-session-media.c:
11307 * gst/rtsp-server/rtsp-session.c:
11308 rtsp-server: added annotations to indicate type of ownership transfer of return values
11309 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11311 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11314 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11316 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11319 * bindings/Makefile.am:
11320 * bindings/vala/Makefile.am:
11321 * bindings/vala/gst-rtsp-server-0.10.deps:
11322 * bindings/vala/gst-rtsp-server-0.10.vapi:
11323 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11324 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11325 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11326 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11327 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11329 bindings: remove vala bindings
11330 They'll be reunited with the other GStreamer bindings
11331 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11333 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11335 * gst/rtsp-server/rtsp-client.c:
11336 * gst/rtsp-server/rtsp-session-media.c:
11337 * gst/rtsp-server/rtsp-session-media.h:
11338 * gst/rtsp-server/rtsp-stream-transport.c:
11339 * gst/rtsp-server/rtsp-stream-transport.h:
11340 rtsp: only create transport when needed
11341 Only create the StreamTransport when configured.
11343 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11345 * gst/rtsp-server/rtsp-client.c:
11346 client: small cleanup
11348 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11350 * gst/rtsp-server/rtsp-client.c:
11351 * gst/rtsp-server/rtsp-client.h:
11352 * gst/rtsp-server/rtsp-stream-transport.c:
11353 * gst/rtsp-server/rtsp-stream-transport.h:
11354 rtsp: refactor configuration of transport
11355 Move the configuration of the transport to a place where it makes
11358 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11360 * gst/rtsp-server/rtsp-client.c:
11361 client: refactor transport parsing
11363 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11365 * gst/rtsp-server/rtsp-client.c:
11366 client: refuse to change the MTU on shared media
11367 If we change the MTU of chared media, it changes for all clients.
11368 We don't want to set the MTU to something large for clients that
11371 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11373 * examples/test-mp4.c:
11374 * gst/rtsp-server/rtsp-media.c:
11375 small fixes to docs and debug
11377 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11379 * gst/rtsp-server/rtsp-stream.c:
11380 stream: transports must already have been removed
11382 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11384 * gst/rtsp-server/rtsp-media.c:
11385 * gst/rtsp-server/rtsp-stream.c:
11386 * gst/rtsp-server/rtsp-stream.h:
11387 stream: improve join and leave of the pipeline
11389 Do the cleanup properly
11392 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11394 * gst/rtsp-server/rtsp-media.c:
11395 media: move unprepare below default implementation
11396 Makes it easier to find the default implementation
11398 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11400 * gst/rtsp-server/rtsp-media.c:
11401 media: signal unprepared when we actually finish
11403 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11405 * gst/rtsp-server/rtsp-media.c:
11406 media: no need to unlock, unprepare does that when needed
11408 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11410 * docs/libs/gst-rtsp-server-sections.txt:
11411 * gst/rtsp-server/rtsp-media-factory.h:
11412 * gst/rtsp-server/rtsp-media-mapping.c:
11413 * gst/rtsp-server/rtsp-media.h:
11414 * gst/rtsp-server/rtsp-params.c:
11415 * gst/rtsp-server/rtsp-server.c:
11416 * gst/rtsp-server/rtsp-session-pool.h:
11417 * gst/rtsp-server/rtsp-session.c:
11418 * gst/rtsp-server/rtsp-session.h:
11419 * gst/rtsp-server/rtsp-stream-transport.h:
11420 * gst/rtsp-server/rtsp-stream.h:
11423 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11425 * gst/rtsp-server/rtsp-client.c:
11426 * gst/rtsp-server/rtsp-media-mapping.h:
11427 * gst/rtsp-server/rtsp-media.c:
11428 * gst/rtsp-server/rtsp-media.h:
11429 * gst/rtsp-server/rtsp-server.h:
11430 * gst/rtsp-server/rtsp-stream.c:
11431 * gst/rtsp-server/rtsp-stream.h:
11432 rtsp: fix MTU setting
11433 Fix setting of the MTU. There is no need for a vmethod.
11435 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11440 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11443 configure: bump version number after refactoring
11445 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11447 * gst/rtsp-server/Makefile.am:
11448 * gst/rtsp-server/rtsp-client.c:
11449 * gst/rtsp-server/rtsp-client.h:
11450 * gst/rtsp-server/rtsp-media-factory-uri.c:
11451 * gst/rtsp-server/rtsp-media-factory.c:
11452 * gst/rtsp-server/rtsp-media-factory.h:
11453 * gst/rtsp-server/rtsp-media.c:
11454 * gst/rtsp-server/rtsp-media.h:
11455 * gst/rtsp-server/rtsp-sdp.c:
11456 * gst/rtsp-server/rtsp-session-media.c:
11457 * gst/rtsp-server/rtsp-session-media.h:
11458 * gst/rtsp-server/rtsp-session.c:
11459 * gst/rtsp-server/rtsp-session.h:
11460 * gst/rtsp-server/rtsp-stream-transport.c:
11461 * gst/rtsp-server/rtsp-stream-transport.h:
11462 * gst/rtsp-server/rtsp-stream.c:
11463 * gst/rtsp-server/rtsp-stream.h:
11464 rtsp: massive refactoring
11465 Make GObjects from the remaining simple structures.
11466 Remove GstRTSPSessionStream, it's not needed.
11467 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11468 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11469 a GstRTSPStream should be transported to a client.
11470 Rename GstRTSPMediaFactory::get_element -> create_element because that
11471 more accurately describes what it does.
11472 Make nice methods instead of poking in the structures.
11473 Move some methods inside the relevant object source code.
11474 Use GPtrArray to store objects instead of plain arrays, it is more
11475 natural and allows us to more easily clean up.
11476 Move the allocation of udp ports to the Stream object. The Stream object
11477 contains the elements needed to stream the media to a client.
11478 Improve the prepare and unprepare methods. Unprepare should now undo
11479 everything prepare did. Improve also async unprepare when doing EOS on
11480 shutdown. Make sure we always unprepare correctly.
11482 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11484 * gst/rtsp-server/rtsp-client.c:
11485 rtsp-client: Unref server address clients connected to
11486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11488 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11490 * gst/rtsp-server/rtsp-server.c:
11491 rtsp-server: don't ref server socket if it is NULL
11492 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11493 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11495 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11497 * tests/check/Makefile.am:
11498 tests: Add libgio link dependency
11499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11501 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11503 * gst/rtsp-server/rtsp-media-mapping.c:
11504 * gst/rtsp-server/rtsp-media-mapping.h:
11505 rtsp-media-mapping: rename find_media vfunc to find_factory
11506 The virtual method and class method should have the same name
11507 so it is correctly represented in GIR file
11508 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11510 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11512 * gst/rtsp-server/rtsp-auth.c:
11513 * gst/rtsp-server/rtsp-client.c:
11514 * gst/rtsp-server/rtsp-media-factory-uri.c:
11515 * gst/rtsp-server/rtsp-media-factory.c:
11516 * gst/rtsp-server/rtsp-media-mapping.c:
11517 * gst/rtsp-server/rtsp-media.c:
11518 * gst/rtsp-server/rtsp-server.c:
11519 * gst/rtsp-server/rtsp-session-pool.c:
11520 * gst/rtsp-server/rtsp-session.c:
11521 rtsp-server: fixed comments and GIR annotations
11522 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11524 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11526 * gst/rtsp-server/rtsp-media-mapping.c:
11527 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11529 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11531 * gst/rtsp-server/rtsp-server.c:
11532 rtsp-server: allow binding on port 0 (binds on a random port)
11534 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11536 * gst/rtsp-server/rtsp-server.c:
11537 * gst/rtsp-server/rtsp-server.h:
11538 rtsp-server: add bound-port property
11539 bound-port can be used to retrieve the port number when the server is bound on
11540 port 0, which binds on a random port.
11542 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11544 * gst/rtsp-server/rtsp-media-factory.c:
11545 * gst/rtsp-server/rtsp-media-factory.h:
11546 rtsp-media-factory: make ::get_element overridable by GI bindings
11547 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11548 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11549 as the invoker for ::get_element(), making it overridable by GI generated
11552 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11554 * gst/rtsp-server/rtsp-media-factory-uri.c:
11555 rtsp-media-factory-uri: don't autoplug parsers in a loop
11556 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11559 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11561 * gst/rtsp-server/Makefile.am:
11562 Explicitly link against gio. Fix link error on mac.
11564 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11566 * gst/rtsp-server/rtsp-session.c:
11567 session: add ttl to the transport header in SETUP
11568 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11570 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11572 * gst/rtsp-server/rtsp-client.c:
11573 * gst/rtsp-server/rtsp-client.h:
11574 * gst/rtsp-server/rtsp-media.c:
11575 client: Use client transport settings for multicast if allowed.
11576 This patch makes it possible for the client to send transport settings for
11577 multicast (destination && ttl). Client settings must be explicitly allowed or
11578 the server will use its own settings.
11579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11581 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11584 Automatic update of common submodule
11585 From 6c0b52c to 6bb6951
11587 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11589 * gst/rtsp-server/rtsp-client.c:
11590 rtsp-client: do not destroy the rtsp watch
11591 Don't destroy the client watch while dispatching. The rtsp watch is
11592 automatically destroyed after the rtsp watch function closed() has
11594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11596 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11599 Automatic update of common submodule
11600 From 4f962f7 to 6c0b52c
11602 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11604 * gst/rtsp-server/rtsp-media.c:
11605 media: fix check for seekability
11607 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11609 * gst/rtsp-server/rtsp-client.c:
11610 client: use more GIO
11611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11613 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11615 * gst/rtsp-server/rtsp-server.c:
11616 server: remove obsolete includes
11618 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11620 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11621 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11622 be available in "on_new_ssrc". The transports are added in
11623 gst_rtsp_media_set_state when going to PLAYING state. However,
11624 "on_new_ssrc" might be called before this happens.
11625 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11627 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11629 * gst/rtsp-server/rtsp-client.c:
11630 * gst/rtsp-server/rtsp-client.h:
11631 rtsp-client: add signals for rtsp requests (fixes #683287)
11633 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11635 * gst/rtsp-server/rtsp-client.c:
11636 * gst/rtsp-server/rtsp-client.h:
11637 add new-session signal to rtsp-client (fixes #683058)
11639 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11642 Automatic update of common submodule
11643 From 668acee to 4f962f7
11645 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11647 * gst/rtsp-server/rtsp-server.c:
11648 * tests/check/gst/rtspserver.c:
11649 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11650 Do not assume that *error is set in g_socket_address_enumerator_next.
11651 Added test_bind_already_in_use unit-test.
11652 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11654 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11657 Automatic update of common submodule
11658 From 94ccf4c to 668acee
11660 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11662 * gst/rtsp-server/rtsp-client.c:
11663 * gst/rtsp-server/rtsp-client.h:
11664 rtsp-client: make create_sdp virtual method
11665 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11667 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11670 Automatic update of common submodule
11671 From 98e386f to 94ccf4c
11673 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11675 * gst/rtsp-server/rtsp-client.c:
11678 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11680 * gst/rtsp-server/rtsp-client.c:
11681 * gst/rtsp-server/rtsp-client.h:
11682 * gst/rtsp-server/rtsp-server.c:
11683 * gst/rtsp-server/rtsp-server.h:
11684 rtsp-server: use an existing socket to establish HTTP tunnel
11685 Make it possible to transfer a socket from an HTTP server to be used as
11686 an RTSP over HTTP tunnel.
11688 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11690 * gst/rtsp-server/rtsp-client.c:
11691 * gst/rtsp-server/rtsp-media.c:
11692 * gst/rtsp-server/rtsp-media.h:
11693 rtsp: Handle the blocksize parameter
11694 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11696 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11698 * tests/check/Makefile.am:
11699 * tests/check/gst/rtspserver.c:
11700 Have unit test get header from source dir, not installed dir
11701 This makes compilation of unit tests work in a build directory other
11702 than the source directory.
11703 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11705 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11707 * gst/rtsp-server/rtsp-media.c:
11708 rtsp-media: update for gst_element_make_from_uri() changes
11710 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11713 * tests/Makefile.am:
11714 * tests/check/Makefile.am:
11715 * tests/check/gst/rtspserver.c:
11716 rtsp: add unit test
11717 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11719 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11721 * gst/rtsp-server/rtsp-media.c:
11722 rtsp-media: don't collect media stats when going to NULL
11723 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11725 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11727 * gst/rtsp-server/rtsp-client.c:
11728 client: don't leak transports
11730 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11732 * gst/rtsp-server/rtsp-client.c:
11733 rtsp-client: free transport on no_stream in SETUP handler
11735 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11737 * gst/rtsp-server/rtsp-client.c:
11738 rtsp-client: changed session media iteration
11739 In client_unlink_session: now don't iterate in session->medias
11740 list where items are removed by gst_rtsp_session_release_media.
11741 Instead, repeatedly remove the first item.
11743 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11745 * gst/rtsp-server/rtsp-client.c:
11746 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11747 GstRTSPSessionMedia is not a GObject type. When the
11748 GstRTSPSession is freed, it will free the media.
11750 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11752 * gst/rtsp-server/rtsp-media-factory.c:
11753 factory: plug pad leak in collect_streams
11754 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11755 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11756 will take one reference, and the other reference will otherwise
11757 give a memory leak.
11759 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11762 configure: suppress some warnings when debug is disabled
11763 Warnings about unused variables should be suppressed if core has the
11764 debug system disabled.
11765 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11767 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11769 * docs/libs/Makefile.am:
11770 docs: fix build in uninstalled setup
11771 Include gst-plugins-base libs properly.
11773 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11775 * docs/libs/gst-rtsp-server.types:
11776 docs: include headers defining rtsp-server object types
11777 Fixes compiler warnings during docs build.
11778 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11780 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11783 configure: Add warning flags for compiler when configuring
11784 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11786 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11789 Automatic update of common submodule
11790 From 03a0e57 to 98e386f
11792 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11795 Automatic update of common submodule
11796 From 1fab359 to 03a0e57
11798 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11800 * gst/rtsp-server/rtsp-client.c:
11801 client: fix GSocketAddress leak in gst_rtsp_client_accept
11802 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11804 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11807 Automatic update of common submodule
11808 From f1b5a96 to 1fab359
11810 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11813 Automatic update of common submodule
11814 From 92b7266 to f1b5a96
11816 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11819 Automatic update of common submodule
11820 From ec1c4a8 to 92b7266
11822 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11825 Automatic update of common submodule
11826 From 3429ba6 to ec1c4a8
11828 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11830 * gst/rtsp-server/rtsp-auth.c:
11831 * gst/rtsp-server/rtsp-client.c:
11832 * gst/rtsp-server/rtsp-media-factory-uri.c:
11833 * gst/rtsp-server/rtsp-server.c:
11834 rtsp: fix compiler warnings
11835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11837 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11840 Automatic update of common submodule
11841 From dc70203 to 3429ba6
11843 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11845 * gst/rtsp-server/rtsp-client.c:
11846 * gst/rtsp-server/rtsp-media-factory.c:
11847 * gst/rtsp-server/rtsp-media-factory.h:
11848 * gst/rtsp-server/rtsp-media.c:
11849 * gst/rtsp-server/rtsp-media.h:
11850 * gst/rtsp-server/rtsp-server.c:
11851 * gst/rtsp-server/rtsp-server.h:
11852 * gst/rtsp-server/rtsp-session-pool.c:
11853 * gst/rtsp-server/rtsp-session-pool.h:
11854 rtsp-server: port to new thread API
11856 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11859 Automatic update of common submodule
11860 From 6db25be to dc70203
11862 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11864 * gst/rtsp-server/rtsp-auth.c:
11865 * gst/rtsp-server/rtsp-auth.h:
11866 * gst/rtsp-server/rtsp-client.c:
11867 rtsp-server: Fix compilation and compiler warnings
11869 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11873 * gst/rtsp-server/Makefile.am:
11874 configure: Modernize autotools setup a bit
11875 Also we now only create tar.bz2 and tar.xz tarballs.
11877 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11880 Automatic update of common submodule
11881 From 464fe15 to 6db25be
11883 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11886 Automatic update of common submodule
11887 From 7fda524 to 464fe15
11889 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11892 * docs/libs/Makefile.am:
11893 * docs/version.entities.in:
11894 * gst-rtsp.spec.in:
11895 * gst/rtsp-server/Makefile.am:
11896 * pkgconfig/Makefile.am:
11897 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11898 * pkgconfig/gstreamer-rtsp-server.pc.in:
11899 * tests/Makefile.am:
11900 rtsp-server: Update versioning
11902 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11904 Merge remote-tracking branch 'origin/0.10'
11906 gst/rtsp-server/rtsp-session-pool.c
11908 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11910 * gst/rtsp-server/rtsp-session-pool.c:
11911 rtsp-server: Don't use deprecated GLib API
11913 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11915 Replace master with 0.11
11917 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11919 Merge branch 'master' into 0.11
11921 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11923 Merge branch 'master' into 0.11
11925 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11928 A couple minor typo fixes
11930 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11932 * gst/rtsp-server/rtsp-media.c:
11933 media: fix state of the appqueue
11935 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11937 * gst/rtsp-server/rtsp-media-factory-uri.c:
11938 factory: use videoconvert
11940 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11942 * gst/rtsp-server/rtsp-media-factory-uri.c:
11943 factory: change to new style caps
11945 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11947 * gst/rtsp-server/rtsp-client.c:
11948 * gst/rtsp-server/rtsp-client.h:
11949 * gst/rtsp-server/rtsp-media-factory-uri.c:
11950 * gst/rtsp-server/rtsp-media.c:
11951 * gst/rtsp-server/rtsp-server.c:
11952 * gst/rtsp-server/rtsp-server.h:
11953 * gst/rtsp-server/rtsp-session-pool.c:
11954 rtsp-server: port to GIO
11957 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11960 configure: fix build
11962 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11965 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11966 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11968 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11971 * examples/Makefile.am:
11972 First rule of gst-rtsp-server club: don't talk about gst-phonon
11974 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11977 * pkgconfig/Makefile.am:
11978 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11979 * pkgconfig/gstreamer-rtsp-server.pc.in:
11980 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11981 For consistency with all other modules.
11983 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11985 * gst/rtsp-server/rtsp-client.c:
11986 rtsp-client: update for new map API
11988 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11991 * bindings/Makefile.am:
11992 * bindings/python/Makefile.am:
11993 * bindings/python/arg-types.py:
11994 * bindings/python/codegen/Makefile.am:
11995 * bindings/python/codegen/__init__.py:
11996 * bindings/python/codegen/argtypes.py:
11997 * bindings/python/codegen/code-coverage.py:
11998 * bindings/python/codegen/codegen.py:
11999 * bindings/python/codegen/definitions.py:
12000 * bindings/python/codegen/defsparser.py:
12001 * bindings/python/codegen/docextract.py:
12002 * bindings/python/codegen/docgen.py:
12003 * bindings/python/codegen/fileprefix.override:
12004 * bindings/python/codegen/fileprefixmodule.c:
12005 * bindings/python/codegen/h2def.py:
12006 * bindings/python/codegen/mergedefs.py:
12007 * bindings/python/codegen/mkskel.py:
12008 * bindings/python/codegen/override.py:
12009 * bindings/python/codegen/reversewrapper.py:
12010 * bindings/python/codegen/scmexpr.py:
12011 * bindings/python/rtspserver-types.defs:
12012 * bindings/python/rtspserver.defs:
12013 * bindings/python/rtspserver.override:
12014 * bindings/python/rtspservermodule.c:
12015 * bindings/python/test.py:
12017 python: remove pygst-based python bindings
12018 pygi is the future, apparently.
12020 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
12023 Automatic update of common submodule
12024 From c463bc0 to 7fda524
12026 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12029 Automatic update of common submodule
12030 From 2a59016 to c463bc0
12032 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12035 Automatic update of common submodule
12036 From 0807187 to 2a59016
12038 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12041 Automatic update of common submodule
12042 From 11f0cd5 to 0807187
12044 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12046 * examples/test-auth.c:
12047 example: update for new caps
12049 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12051 * examples/test-video.c:
12052 * gst/rtsp-server/rtsp-client.c:
12053 * gst/rtsp-server/rtsp-media-factory-uri.c:
12054 * gst/rtsp-server/rtsp-media.c:
12055 * gst/rtsp-server/rtsp-media.h:
12056 * gst/rtsp-server/rtsp-session.c:
12057 * gst/rtsp-server/rtsp-session.h:
12058 rtsp-server: port some more to 0.11
12060 Remove bufferlist stuff
12061 Update for new API.
12062 Add queue before appsink now that preroll-queue-len is gone.
12063 Update for request pad changes.
12065 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12067 Merge branch 'master' into 0.11
12069 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12071 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12072 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12073 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12075 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12077 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12078 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12079 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12081 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12083 Merge branch 'master' into 0.11
12085 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12087 * gst/rtsp-server/rtsp-media.c:
12088 * gst/rtsp-server/rtsp-media.h:
12089 media: add a seekable boolean
12090 Maintain the seekable state with a new variable instead of reusing the
12093 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
12095 * gst/rtsp-server/rtsp-media.c:
12096 Disallow seek in live media
12098 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12100 Merge branch 'master' into 0.11
12102 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
12104 * gst/rtsp-server/rtsp-server.c:
12105 #ifdef statements for windows socket creation were missing
12107 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
12110 Automatic update of common submodule
12111 From a39eb83 to 11f0cd5
12113 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
12116 Automatic update of common submodule
12117 From 605cd9a to a39eb83
12119 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12121 Merge branch 'master' into 0.11
12123 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12125 * gst/rtsp-server/rtsp-client.c:
12126 client: use method to access property
12128 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12130 * gst/rtsp-server/rtsp-media-factory.c:
12131 * gst/rtsp-server/rtsp-media-factory.h:
12132 media-factory: add protocols property
12133 Add a property to configure the allowed protocols in the media created from the
12136 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12138 * gst/rtsp-server/rtsp-media-factory.c:
12139 * gst/rtsp-server/rtsp-media-factory.h:
12140 media-factory: add media-configure signal
12141 Add signal to allow the application to configure the media after it was created
12144 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12146 * gst/rtsp-server/rtsp-client.c:
12147 client: use method to access property
12149 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12151 * gst/rtsp-server/rtsp-media-factory.c:
12152 * gst/rtsp-server/rtsp-media-factory.h:
12153 media-factory: add protocols property
12154 Add a property to configure the allowed protocols in the media created from the
12157 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12159 * gst/rtsp-server/rtsp-media-factory.c:
12160 * gst/rtsp-server/rtsp-media-factory.h:
12161 media-factory: add media-configure signal
12162 Add signal to allow the application to configure the media after it was created
12165 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12167 Merge branch 'master' into 0.11
12169 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12171 * gst/rtsp-server/rtsp-client.c:
12172 client: use media multicast group
12174 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12176 * gst/rtsp-server/rtsp-media-factory.h:
12177 * gst/rtsp-server/rtsp-server.h:
12178 * gst/rtsp-server/rtsp-session-pool.h:
12179 * gst/rtsp-server/rtsp-session.h:
12182 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12184 * gst/rtsp-server/rtsp-client.c:
12185 * gst/rtsp-server/rtsp-sdp.h:
12186 sdp: copy and free the server ip address
12187 Copy and free the server ip address to make memory management easier later.
12189 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12191 * gst/rtsp-server/rtsp-media-factory.c:
12192 media-factory: configure multicast in media
12194 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12196 * gst/rtsp-server/rtsp-media.c:
12197 * gst/rtsp-server/rtsp-media.h:
12198 media: add property for multicast group
12199 Add a property to configure the multicast group in the media.
12200 Based on patches from Marc Leeman and Robert Krakora.
12202 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12204 * gst/rtsp-server/rtsp-media-factory.c:
12205 * gst/rtsp-server/rtsp-media-factory.h:
12206 media-factory: add property for multicast group
12207 Add a property to configure the multicast group in the media factory.
12208 Based on patches from Marc Leeman and Robert Krakora.
12210 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12212 * gst/rtsp-server/rtsp-client.c:
12213 client: do configuration of transport in one place
12214 Move the configuration of the transport destination address to where we also
12215 configure the other bits.
12217 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12219 * gst/rtsp-server/rtsp-client.c:
12220 client: use media multicast group
12222 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12224 * gst/rtsp-server/rtsp-media-factory.h:
12225 * gst/rtsp-server/rtsp-server.h:
12226 * gst/rtsp-server/rtsp-session-pool.h:
12227 * gst/rtsp-server/rtsp-session.h:
12230 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12232 * gst/rtsp-server/rtsp-client.c:
12233 * gst/rtsp-server/rtsp-sdp.h:
12234 sdp: copy and free the server ip address
12235 Copy and free the server ip address to make memory management easier later.
12237 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12239 * gst/rtsp-server/rtsp-media-factory.c:
12240 media-factory: configure multicast in media
12242 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12244 * gst/rtsp-server/rtsp-media.c:
12245 * gst/rtsp-server/rtsp-media.h:
12246 media: add property for multicast group
12247 Add a property to configure the multicast group in the media.
12248 Based on patches from Marc Leeman and Robert Krakora.
12250 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12252 * gst/rtsp-server/rtsp-media-factory.c:
12253 * gst/rtsp-server/rtsp-media-factory.h:
12254 media-factory: add property for multicast group
12255 Add a property to configure the multicast group in the media factory.
12256 Based on patches from Marc Leeman and Robert Krakora.
12258 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12260 * gst/rtsp-server/rtsp-client.c:
12261 client: do configuration of transport in one place
12262 Move the configuration of the transport destination address to where we also
12263 configure the other bits.
12265 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12267 Merge branch 'master' into 0.11
12269 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12271 * gst/rtsp-server/rtsp-client.c:
12272 client: destroy pipeline on client disconnect with no prior TEARDOWN.
12273 The problem occurs when the client abruptly closes the connection without
12274 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
12275 server is where the pipeline gets torn down. Since this handler is not called,
12276 the pipeline remains and is up and running. Subsequent clients get their own
12277 pipelines and if the do not issue TEARDOWNs then those pipelines will also
12278 remain up and running. This is a resource leak.
12280 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12282 Merge branch 'master' into 0.11
12284 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
12286 * gst/rtsp-server/rtsp-media-factory.c:
12287 * gst/rtsp-server/rtsp-media-factory.h:
12288 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
12289 For example, it can be used to retrieve source elements like appsrc, in a more
12290 convenient way than subclassing get_element.
12292 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12294 Merge branch 'master' into 0.11
12296 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
12298 * gst/rtsp-server/rtsp-server.c:
12299 rtsp-server: hold on to reference while using object
12301 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12303 * gst/rtsp-server/rtsp-media.c:
12306 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12309 configure: use unstable api
12311 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12313 * gst/rtsp-server/rtsp-client.c:
12314 client: fix reference counting
12316 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12318 * gst/rtsp-server/rtsp-client.c:
12319 * gst/rtsp-server/rtsp-media.c:
12320 fix compiler warnings about unused variables
12322 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12324 * examples/test-launch.c:
12325 * examples/test-readme.c:
12326 * examples/test-uri.c:
12327 * examples/test-video.c:
12328 examples: tell rtsp uri when ready
12330 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12333 Automatic update of common submodule
12334 From 69b981f to 605cd9a
12336 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12338 * gst/rtsp-server/rtsp-client.c:
12339 client: update for buffer API change
12341 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12343 * gst/rtsp-server/Makefile.am:
12344 Makefile.am: 0.10 => @GST_MAJORMINOR@
12346 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12348 * gst/rtsp-server/rtsp-media-factory-uri.c:
12349 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12351 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12353 * gst/rtsp-server/.gitignore:
12354 .gitignore: 0.10 => 0.11
12356 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12358 * gst/rtsp-server/Makefile.am:
12359 Makefile.am: 0.10 => @GST_MAJORMINOR@
12361 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12363 Merge branch 'master' into 0.11
12365 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12368 Automatic update of common submodule
12369 From 9e5bbd5 to 69b981f
12371 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12374 Automatic update of common submodule
12375 From fd35073 to 9e5bbd5
12377 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12380 Automatic update of common submodule
12381 From 46dfcea to fd35073
12383 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12385 * gst/rtsp-server/rtsp-media-factory-uri.c:
12386 * gst/rtsp-server/rtsp-media.c:
12387 media: port to new caps API
12389 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12391 Merge branch 'master' into 0.11
12393 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12395 * bindings/vala/gst-rtsp-server-0.10.vapi:
12396 Updated Vala bindings.
12397 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12399 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12401 * gst/rtsp-server/rtsp-server.c:
12402 * gst/rtsp-server/rtsp-server.h:
12403 Add a signal for newly connected clients.
12404 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12406 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12408 * bindings/python/rtspserver.override:
12409 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12411 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12413 * gst/rtsp-server/Makefile.am:
12414 * gst/rtsp-server/rtsp-client.c:
12415 * gst/rtsp-server/rtsp-funnel.c:
12416 * gst/rtsp-server/rtsp-funnel.h:
12417 * gst/rtsp-server/rtsp-media.c:
12418 rtsp-server: port to 0.11
12420 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12425 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12427 Merge branch 'master' into 0.11
12432 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12435 Automatic update of common submodule
12436 From c3cafe1 to 46dfcea
12438 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12440 * bindings/python/Makefile.am:
12441 * bindings/python/rtspserver.defs:
12442 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12444 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12446 * bindings/python/arg-types.py:
12447 python bindings: add GstRTSPUrlParam
12448 Needed to implement MediaFactory virtual proxies
12450 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12452 * bindings/python/arg-types.py:
12453 python bindings: fix returning GstRTSPUrl types
12455 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12457 * bindings/python/arg-types.py:
12458 python bindings: add arg type for GstRTSPUrl
12460 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12462 * bindings/python/rtspserver.defs:
12463 python bindings: fix the definition of MediaFactory.collect_stream
12465 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12468 Automatic update of common submodule
12469 From 1ccbe09 to c3cafe1
12471 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12474 Automatic update of common submodule
12475 From 193b717 to 1ccbe09
12477 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12480 Automatic update of common submodule
12481 From b77e2bf to 193b717
12483 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12486 build: Include lcov.mak to allow test coverage report generation
12488 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12491 Automatic update of common submodule
12492 From d8814b6 to b77e2bf
12494 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12497 Automatic update of common submodule
12498 From 6aaa286 to d8814b6
12500 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12503 Automatic update of common submodule
12504 From 6aec6b9 to 6aaa286
12506 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12509 autogen: wingo signed comment
12511 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12513 * gst/rtsp-server/rtsp-session-pool.c:
12514 session: use full charset for RTSP session ID
12515 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12516 session ID more difficult.
12517 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12519 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12521 * gst/rtsp-server/Makefile.am:
12522 rtsp-server: Don't install the funnel header
12524 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12527 Automatic update of common submodule
12528 From 1de7f6a to 6aec6b9
12530 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12533 configure: require core/base 0.10.31
12534 Needed at least for gst_plugin_feature_rank_compare_func().
12536 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12539 Automatic update of common submodule
12540 From f94d739 to 1de7f6a
12542 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12544 * gst/rtsp-server/rtsp-media.c:
12545 media: remove more unused code
12547 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12549 * gst/rtsp-server/rtsp-media.c:
12550 * gst/rtsp-server/rtsp-media.h:
12551 media: remove duplicate filtering
12552 Remove the duplicate filtering code now that we have a released -good version.
12553 Give a warning instead.
12555 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12557 * gst/rtsp-server/rtsp-media-factory.c:
12558 * gst/rtsp-server/rtsp-media.c:
12559 media: fix default buffer size
12561 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12563 * gst/rtsp-server/rtsp-media-factory.c:
12564 * gst/rtsp-server/rtsp-media-factory.h:
12565 media-factory: add property to configure the buffer-size
12566 Add a property to configure the kernel UDP buffer size.
12568 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12570 * gst/rtsp-server/rtsp-media.c:
12571 * gst/rtsp-server/rtsp-media.h:
12572 media: add property to configure kernel buffer sizes
12573 Add a property to configure the kernel UDP buffer size.
12575 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12578 configure: set PYGOBJECT_REQ before using it
12579 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12581 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12583 * docs/Makefile.am:
12584 docs: recursive into sub-directories on 'make upload'
12586 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12588 * docs/libs/gst-rtsp-server-docs.sgml:
12589 * docs/version.entities.in:
12590 docs: mention full version these docs are for, not just major-minor
12592 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12595 back to development
12597 === release 0.10.8 ===
12599 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12604 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12606 * gst/rtsp-server/rtsp-server.c:
12607 rtsp-server: clarify docs a little
12609 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12611 * gst/rtsp-server/rtsp-media.c:
12612 media: init debug category before starting thread
12614 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12616 * gst/rtsp-server/rtsp-auth.c:
12617 auth: add realm to make it more spec compliant
12619 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12621 * gst/rtsp-server/rtsp-server.c:
12622 * gst/rtsp-server/rtsp-server.h:
12623 server: add locking
12625 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12627 * examples/test-video.c:
12628 example: improve example docs a little
12630 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12632 * gst/rtsp-server/rtsp-server.c:
12633 server: ensure the watch has a ref to the server
12635 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12637 * gst/rtsp-server/rtsp-server.c:
12638 server: simpify channel function
12640 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12642 * gst/rtsp-server/rtsp-server.c:
12643 * gst/rtsp-server/rtsp-server.h:
12644 server: simplify management of channel and source
12645 We don't need to keep around the channel and source objects. Let the mainloop
12646 and the source manage the source and channel respectively.
12648 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12654 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12656 * tests/.gitignore:
12657 * tests/Makefile.am:
12658 * tests/test-cleanup.c:
12659 tests: add tests directory and cleanup test
12661 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12663 * gst/rtsp-server/rtsp-media-factory-uri.c:
12664 * gst/rtsp-server/rtsp-media-factory.c:
12665 * gst/rtsp-server/rtsp-media-mapping.c:
12666 * gst/rtsp-server/rtsp-media.c:
12667 * gst/rtsp-server/rtsp-session-pool.c:
12668 * gst/rtsp-server/rtsp-session.c:
12669 server: improve debugging in various objects
12671 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12673 * gst/rtsp-server/rtsp-server.c:
12674 server: chain up to the parent finalize
12676 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12678 * bindings/python/rtspserver-types.defs:
12679 * bindings/python/rtspserver.defs:
12680 * bindings/python/rtspserver.override:
12681 * bindings/python/test.py:
12682 gst-rtsp-server: update python bindings
12684 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12686 * gst/rtsp-server/rtsp-client.c:
12687 client: use the response from the clientstate
12688 Create the response object only once and store in the client state.
12689 Make all methods use the state response,
12691 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12693 * gst/rtsp-server/rtsp-server.c:
12694 server: use signal to keep track of clients
12695 Keep track of all the clients that the server creates and remove them when they
12696 fire the 'closed' signal.
12698 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12700 * gst/rtsp-server/rtsp-client.c:
12701 * gst/rtsp-server/rtsp-client.h:
12702 client: emit signal when closing
12704 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12706 * examples/.gitignore:
12707 * examples/Makefile.am:
12708 * examples/test-auth.c:
12709 * examples/test-video.c:
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-media-factory.c:
12714 * gst/rtsp-server/rtsp-media.c:
12715 * gst/rtsp-server/rtsp-media.h:
12716 * gst/rtsp-server/rtsp-session-pool.h:
12717 * gst/rtsp-server/rtsp-session.h:
12718 media: enable per factory authorisations
12719 Allow for adding a GstRTSPAuth on the factory and media level and check
12720 permissions when accessing the factory.
12721 Add hints to the auth methods for future more fine grained authorisation.
12722 Add example application for per factory authentication.
12724 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12726 * gst/rtsp-server/rtsp-auth.c:
12727 * gst/rtsp-server/rtsp-auth.h:
12728 * gst/rtsp-server/rtsp-client.c:
12729 * gst/rtsp-server/rtsp-client.h:
12730 * gst/rtsp-server/rtsp-params.c:
12731 * gst/rtsp-server/rtsp-params.h:
12732 rtsp-server: Pass ClientState structure arround
12733 Pass the collected information for the ongoing request in a GstRTSPClientState
12734 structure that we can then pass around to simplify the method arguments. This
12735 will also be handy when we implement logging functionality.
12737 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12739 * gst/rtsp-server/rtsp-media-factory.c:
12740 * gst/rtsp-server/rtsp-media-factory.h:
12741 media-factory: add methods to configure authorisation
12743 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12745 * gst/rtsp-server/rtsp-client.c:
12746 client: unref auth in finalize
12748 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12750 * gst/rtsp-server/rtsp-server.c:
12751 server: unref auth in finalize
12753 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12755 * docs/libs/gst-rtsp-server-docs.sgml:
12756 * docs/libs/gst-rtsp-server-sections.txt:
12757 * docs/libs/gst-rtsp-server.types:
12758 docs: add more docs
12760 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12762 * gst/rtsp-server/rtsp-server.c:
12763 * gst/rtsp-server/rtsp-server.h:
12764 server: separate create and accept
12765 Create separate create and accept methods so that subclasses can create custom
12767 Configure the server in the client object and prepare for keeping track of
12770 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12772 * gst/rtsp-server/rtsp-client.c:
12773 * gst/rtsp-server/rtsp-client.h:
12774 client: add support for setting the server.
12775 Add support for keeping a ref to the server that started this client
12778 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12780 * gst/rtsp-server/rtsp-auth.c:
12781 auth: fix memleak and add some docs
12782 Fix a memleak of the basic auth token.
12783 Add docs for the helper function
12785 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12787 * gst/rtsp-server/rtsp-auth.c:
12788 * gst/rtsp-server/rtsp-auth.h:
12789 * gst/rtsp-server/rtsp-client.c:
12790 client: delegate setup of auth to the manager
12791 Delegate the configuration of the authentication tokens to the manager object
12794 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12796 * examples/test-video.c:
12797 * gst/rtsp-server/Makefile.am:
12798 * gst/rtsp-server/rtsp-auth.c:
12799 * gst/rtsp-server/rtsp-auth.h:
12800 * gst/rtsp-server/rtsp-client.c:
12801 * gst/rtsp-server/rtsp-client.h:
12802 * gst/rtsp-server/rtsp-server.c:
12803 * gst/rtsp-server/rtsp-server.h:
12804 auth: add authentication object
12805 Add an object that can check the authorization of requests.
12806 Implement basic authentication.
12807 Add example authentication to test-video
12809 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12811 * gst/rtsp-server/rtsp-server.c:
12812 * gst/rtsp-server/rtsp-server.h:
12813 server: move includes back
12814 the includes are needed for sockaddr_in.
12816 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12818 * gst/rtsp-server/rtsp-client.c:
12819 * gst/rtsp-server/rtsp-client.h:
12820 * gst/rtsp-server/rtsp-server.c:
12821 * gst/rtsp-server/rtsp-server.h:
12822 rtsp: move network includes where they are needed
12824 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12826 * gst/rtsp-server/rtsp-media.h:
12827 rtsp-media.h: Minor corrections in comments.
12830 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12833 Automatic update of common submodule
12834 From e572c87 to f94d739
12836 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12840 * docs/libs/.gitignore:
12841 * examples/.gitignore:
12842 * gst/rtsp-server/.gitignore:
12845 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12847 * docs/libs/Makefile.am:
12848 docs: We don't build ps/pdf for API reference docs
12850 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12853 Automatic update of common submodule
12854 From ccbaa85 to e572c87
12856 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12859 Automatic update of common submodule
12860 From 46445ad to ccbaa85
12862 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12864 * gst/rtsp-server/Makefile.am:
12865 * gst/rtsp-server/rtsp-funnel.c:
12866 * gst/rtsp-server/rtsp-funnel.h:
12867 * gst/rtsp-server/rtsp-media.c:
12868 funnel: rename fsfunnel to rtspfunnel
12869 Rename the funnel to avoid conflicts with the farsight one.
12871 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12873 * gst/rtsp-server/Makefile.am:
12874 * gst/rtsp-server/fs-funnel.c:
12875 * gst/rtsp-server/fs-funnel.h:
12876 * gst/rtsp-server/rtsp-media.c:
12877 rtsp-media: add and use fsfunnel
12878 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12879 select-all property that we need.
12881 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12883 * gst/rtsp-server/Makefile.am:
12884 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12885 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12886 for the g-ir-compiler, rather than just assuming the env var has
12889 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12896 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12898 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12901 * gst/rtsp-server/Makefile.am:
12902 gobject-introspection: fix g-i build for uninstalled setup
12903 Requires gst-plugins-base git (> 0.10.31.2).
12905 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12907 * examples/test-uri.c:
12908 examples: add some more options and comments
12910 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12912 * gst/rtsp-server/rtsp-media-factory-uri.c:
12913 factory-uri: use right property type
12915 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12917 * gst/rtsp-server/rtsp-media-factory-uri.c:
12918 factory-uri: attempt to configure buffer-lists
12919 Attempt to configure buffer lists in the payloader for improved performance.
12921 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12923 * gst/rtsp-server/rtsp-media.c:
12924 media: attempt to configure bigger UDP buffers
12925 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12926 send buffers with high bitrate streams.
12928 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12930 * gst/rtsp-server/rtsp-client.c:
12931 client: use the socket length from getsockname
12932 Use the length returned by getsockname to perform the getnameinfo call because
12933 the size can depend on the socket type and platform.
12936 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12938 * docs/libs/gst-rtsp-server-docs.sgml:
12939 * docs/libs/gst-rtsp-server-sections.txt:
12940 docs: add uri factory to the docs
12942 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12944 * gst/rtsp-server/rtsp-client.c:
12945 * gst/rtsp-server/rtsp-media.h:
12948 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12950 * gst/rtsp-server/rtsp-client.c:
12951 * gst/rtsp-server/rtsp-media.c:
12952 * gst/rtsp-server/rtsp-media.h:
12953 * gst/rtsp-server/rtsp-session.c:
12954 * gst/rtsp-server/rtsp-session.h:
12955 rtsp-server: add support for buffer lists
12956 Add support for sending bufferlists received from appsink.
12959 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12961 * gst/rtsp-server/rtsp-client.c:
12962 * gst/rtsp-server/rtsp-media.c:
12963 * gst/rtsp-server/rtsp-media.h:
12964 * gst/rtsp-server/rtsp-sdp.c:
12965 media: make method to retrieve the play range
12966 Make a method to retrieve the playback range so that we can conditionally create
12967 a different range for the SDP and the PLAY requests.
12969 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12971 * gst/rtsp-server/rtsp-media.c:
12972 * gst/rtsp-server/rtsp-media.h:
12973 media: add signal to notify of state changes
12975 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12977 * gst/rtsp-server/rtsp-client.h:
12978 client: cleanup headers
12980 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12982 * gst/rtsp-server/rtsp-client.c:
12985 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12987 * gst/rtsp-server/rtsp-media-factory-uri.c:
12988 * gst/rtsp-server/rtsp-media-factory-uri.h:
12989 factory-uri: add support for gstpay
12990 Add an option to prefer gstpay over decoder + raw payloader.
12992 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12994 * gst/rtsp-server/rtsp-media-factory-uri.c:
12995 * gst/rtsp-server/rtsp-media-factory-uri.h:
12996 factory-uri: rework the autoplugger.
12997 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
13000 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13002 * gst/rtsp-server/rtsp-media-factory-uri.c:
13003 factory-uri: use better factory filter
13004 Make better payloader filter based on autoplug rank and RTP use case.
13006 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13009 Automatic update of common submodule
13010 From 169462a to 46445ad
13012 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13014 * gst/rtsp-server/rtsp-server.c:
13015 server: set SO_REUSEADDR before bind
13016 Set the SO_REUSEADDR _before_ bind() to make it actually work.
13018 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13020 * gst/rtsp-server/rtsp-media.c:
13021 * gst/rtsp-server/rtsp-media.h:
13022 media: emit prepared signal when prepared
13023 Make a 'prepared' signal and emit it when we successfully prepared the element.
13024 This signal can be used to configure the media object after it has been prepared
13027 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
13030 Automatic update of common submodule
13031 From 011bcc8 to 169462a
13033 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
13035 python an optional dependency
13036 * configure.ac: Move up valgrind and g-i checks. Make the python
13037 dependency optional, as it was before.
13039 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13041 Merge branch 'master' into 0.11
13046 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13048 * gst/rtsp-server/rtsp-media.c:
13049 media: update range when active clients changed
13050 When we changed the number of active clients, update the current range
13051 information because we want the second client connecting to a shared resource
13052 continue from where the stream currently.
13054 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13056 * gst/rtsp-server/rtsp-media-factory-uri.c:
13057 * gst/rtsp-server/rtsp-media-factory-uri.h:
13058 factory-uri: add colorspace and fix pt
13059 Rework the way we pass data to the autoplugger.
13060 When we have raw caps, plug a converter element to make pluggin to raw
13061 payloaders more successful.
13062 Make sure all dynamically plugged payloaders have a unique payload types.
13064 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13066 * examples/Makefile.am:
13067 * examples/test-uri.c:
13068 example: add example of the uri factory
13070 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13072 * gst/rtsp-server/Makefile.am:
13073 * gst/rtsp-server/rtsp-media-factory-uri.c:
13074 * gst/rtsp-server/rtsp-media-factory-uri.h:
13075 * gst/rtsp-server/rtsp-server.h:
13076 factory-uri: add a factory to stream any URI
13077 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
13080 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13082 * gst/rtsp-server/rtsp-media.c:
13083 * gst/rtsp-server/rtsp-media.h:
13084 media: ignore spurious ASYNC_DONE messages
13085 When we are dynamically adding pads, the addition of the udpsrc elements will
13086 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
13087 the real ASYNC_DONE when everything is prerolled.
13089 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13091 * gst/rtsp-server/rtsp-media-factory.c:
13092 * gst/rtsp-server/rtsp-media-factory.h:
13093 media-factory: make lock macro
13095 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
13097 * gst/rtsp-server/rtsp-client.c:
13098 rtsp-server: Remove unused variable and dead assignment
13100 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
13102 * examples/test-launch.c:
13103 * examples/test-mp4.c:
13104 * examples/test-ogg.c:
13105 * examples/test-readme.c:
13106 * examples/test-sdp.c:
13107 * examples/test-video.c:
13108 examples: Run gst-indent
13110 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
13112 * gst/rtsp-server/rtsp-client.c:
13113 * gst/rtsp-server/rtsp-media-factory.c:
13114 * gst/rtsp-server/rtsp-media-mapping.c:
13115 * gst/rtsp-server/rtsp-media.c:
13116 * gst/rtsp-server/rtsp-params.c:
13117 * gst/rtsp-server/rtsp-sdp.c:
13118 * gst/rtsp-server/rtsp-server.c:
13119 * gst/rtsp-server/rtsp-session-pool.c:
13120 * gst/rtsp-server/rtsp-session.c:
13121 rtsp-server: Run gst-indent
13122 Since it wasn't using the upstream common previously, there was no
13123 indentation check before commiting.
13125 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
13127 * gst/rtsp-server/rtsp-media-mapping.h:
13128 * gst/rtsp-server/rtsp-media.c:
13129 * gst/rtsp-server/rtsp-media.h:
13130 * gst/rtsp-server/rtsp-sdp.c:
13131 * gst/rtsp-server/rtsp-session-pool.h:
13132 * gst/rtsp-server/rtsp-session.c:
13133 * gst/rtsp-server/rtsp-session.h:
13134 rtsp-server: Some more doc fixups
13136 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13139 Makefile: Add cruft-cleaning support
13141 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13145 * docs/Makefile.am:
13146 * docs/libs/Makefile.am:
13147 * docs/libs/gst-rtsp-server-docs.sgml:
13148 * docs/libs/gst-rtsp-server-sections.txt:
13149 * docs/libs/gst-rtsp-server.types:
13150 * docs/version.entities.in:
13151 docs: Add gtk-doc build system
13153 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13155 * gst/rtsp-server/Makefile.am:
13156 Makefile.am: Use standard GIR make behaviour
13158 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13162 autogen/configure: Bring more in sync to standard gst module behaviour
13164 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13166 * gst/rtsp-server/rtsp-media.c:
13167 media: warn and fail when gstrtpbin is not found
13169 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13172 configure: open 0.11 branch
13174 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
13178 Add common submodule
13180 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
13182 * common/ChangeLog:
13183 * common/Makefile.am:
13184 * common/c-to-xml.py:
13185 * common/check.mak:
13186 * common/coverage/coverage-report-entry.pl:
13187 * common/coverage/coverage-report.pl:
13188 * common/coverage/coverage-report.xsl:
13189 * common/coverage/lcov.mak:
13190 * common/gettext.patch:
13191 * common/glib-gen.mak:
13192 * common/gst-autogen.sh:
13193 * common/gst-xmlinspect.py:
13195 * common/gstdoc-scangobj:
13196 * common/gtk-doc-plugins.mak:
13197 * common/gtk-doc.mak:
13198 * common/m4/.gitignore:
13199 * common/m4/Makefile.am:
13200 * common/m4/README:
13201 * common/m4/as-ac-expand.m4:
13202 * common/m4/as-auto-alt.m4:
13203 * common/m4/as-compiler-flag.m4:
13204 * common/m4/as-compiler.m4:
13205 * common/m4/as-docbook.m4:
13206 * common/m4/as-libtool-tags.m4:
13207 * common/m4/as-libtool.m4:
13208 * common/m4/as-python.m4:
13209 * common/m4/as-scrub-include.m4:
13210 * common/m4/as-version.m4:
13211 * common/m4/ax_create_stdint_h.m4:
13212 * common/m4/check.m4:
13213 * common/m4/glib-gettext.m4:
13214 * common/m4/gst-arch.m4:
13215 * common/m4/gst-args.m4:
13216 * common/m4/gst-check.m4:
13217 * common/m4/gst-debuginfo.m4:
13218 * common/m4/gst-default.m4:
13219 * common/m4/gst-doc.m4:
13220 * common/m4/gst-error.m4:
13221 * common/m4/gst-feature.m4:
13222 * common/m4/gst-function.m4:
13223 * common/m4/gst-gettext.m4:
13224 * common/m4/gst-glib2.m4:
13225 * common/m4/gst-libxml2.m4:
13226 * common/m4/gst-plugindir.m4:
13227 * common/m4/gst-valgrind.m4:
13228 * common/m4/gtk-doc.m4:
13229 * common/m4/introspection.m4:
13230 * common/m4/pkg.m4:
13231 * common/mangle-tmpl.py:
13232 * common/plugins.xsl:
13234 * common/release.mak:
13235 * common/scangobj-merge.py:
13236 * common/upload.mak:
13237 common: Remove static version
13239 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
13241 * common/m4/introspection.m4:
13242 Update introspection.m4 to match usage
13244 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13248 Remove old stuff from the README
13250 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13253 back to development
13255 === release 0.10.7 ===
13257 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13262 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13264 * examples/test-ogg.c:
13265 test-ogg: remove parsers
13266 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
13267 buffers with timestamps. Using the parsers also seems to break things.
13269 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13271 * bindings/vala/gst-rtsp-server-0.10.vapi:
13272 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13273 Updated Vala bindings
13275 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13277 * common/m4/introspection.m4:
13279 * gst/rtsp-server/Makefile.am:
13280 Added initial gobject-introspection support
13282 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13284 * gst/rtsp-server/rtsp-media-factory.c:
13285 media-factory: don't use host for shared hash key
13286 When we generate the key to share made between connections, don't include the
13287 host used to connect so that we can share media even if between clients that
13288 connected with localhost and ones with the ip address.
13290 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13292 * bindings/vala/Makefile.am:
13293 build: fix distcheck
13295 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13297 * bindings/vala/gst-rtsp-server-0.10.vapi:
13298 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13299 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13300 Update Vala bindings
13302 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13304 * bindings/vala/Makefile.am:
13306 Fix configure checks and installation location for Vala bindings
13309 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13312 back to development
13314 === release 0.10.6 ===
13316 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13319 configure: release 0.10.6
13321 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13323 * gst/rtsp-server/rtsp-media.c:
13324 media: help the compiler a little
13326 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13328 * gst/rtsp-server/rtsp-media.c:
13329 * gst/rtsp-server/rtsp-media.h:
13330 * gst/rtsp-server/rtsp-session.c:
13331 media: cleanup media transport before freeing
13332 Cleanup the media transport data before freeing. In particular, remove the qdata
13333 from the rtpsource object.
13335 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13337 * gst/rtsp-server/rtsp-media-factory.c:
13338 * gst/rtsp-server/rtsp-media-factory.h:
13339 * gst/rtsp-server/rtsp-media.c:
13340 * gst/rtsp-server/rtsp-media.h:
13341 media-factory: add eos-shutdown property
13342 Add an eos-shutdown property that will send an EOS to the pipeline before
13343 shutting it down. This allows for nice cleanup in case of a muxer.
13346 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13348 * gst/rtsp-server/rtsp-media.c:
13349 * gst/rtsp-server/rtsp-media.h:
13350 media: use multiudpsink send-duplicates when we can
13351 If we have a new enough multiudpsink with the send-duplicates property, use this
13352 instead of doing our own filtering. Our custom filtering code should eventually
13353 be removed when we can depend on a released -good.
13355 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13357 * gst/rtsp-server/rtsp-media.c:
13358 media: don't leak destinations
13359 Refactor and cleanup the destinations array when the stream is destroyed.
13361 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13363 * gst/rtsp-server/rtsp-media.c:
13364 * gst/rtsp-server/rtsp-media.h:
13365 media: don't add udp addresses multiple times
13366 Keep track of the udp addresses we added to udpsink and never add the same udp
13367 destination twice. This avoids duplicate packets when using multicast.
13369 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13371 * gst/rtsp-server/rtsp-server.c:
13372 server: disable use of SO_LINGER
13373 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13374 server close()s the connection.
13376 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13378 * gst/rtsp-server/rtsp-server.c:
13379 server: use 5 second linger period in SO_LINGER
13380 Wait 5 seconds before clearing the send buffers and reseting the connection with
13381 the client when we do a close. This should be enough time to get the message to
13385 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13387 * gst/rtsp-server/rtsp-server.c:
13388 server: use SO_LINGER
13389 SO_LINGER on the socket will make sure that any pending data on the socket is
13390 flushed ASAP and that the socket connection is reset. This makes sure that the
13391 socket can be reused immediately.
13394 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13397 README: add blurb about shared media factories
13399 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13401 * gst/rtsp-server/rtsp-media.c:
13402 Add stdlib.h for atoi()
13404 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13406 * bindings/python/Makefile.am:
13407 * bindings/vala/Makefile.am:
13408 build: distcheck fixes
13409 Fix 'make distcheck', somewhat (it still fails because it tries to
13410 install files into /usr/share/vala/vapi/ irrespective of the
13411 configured prefix).
13413 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13416 configure: bump core/base requirements to released version
13417 Makes things less confusing for people.
13419 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13422 configure: fail if GStreamer core/base requirements are not met
13424 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13426 * gst/rtsp-server/rtsp-client.c:
13427 client: improve client cleanups
13428 Make sure the session does not timeout when using TCP. We need to do this
13429 because quicktime player does not send RTCP for some reason in tunneled
13431 Refactor some cleanup code.
13434 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13436 * gst/rtsp-server/rtsp-session.c:
13437 * gst/rtsp-server/rtsp-session.h:
13438 session: add support for prevent session timeouts
13439 Add an atomix counter to prevent session timeouts when we are, for example,
13440 streaming over TCP.
13442 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13444 * gst/rtsp-server/rtsp-client.c:
13445 client: fix unlink on session timeouts
13446 When our session times out, make sure we unlink all streams in this
13448 Remove the tunnelid when closing the connection.
13450 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13452 * gst/rtsp-server/rtsp-session.c:
13453 session: small cleanups
13455 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13457 * gst/rtsp-server/rtsp-client.c:
13458 client: handle lost_tunnel callbacks
13459 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13460 hashtable so that we can reuse it for when the client reopens the POST
13462 Close the connection after a TEARDOWN.
13463 Make sure or watchid is cleared when the watch is removed.
13466 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13468 * gst/rtsp-server/rtsp-client.c:
13469 * gst/rtsp-server/rtsp-media.c:
13470 * gst/rtsp-server/rtsp-sdp.c:
13471 rtsp-server: add more support for multicast
13473 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13476 * gst/rtsp-server/rtsp-media.c:
13477 * gst/rtsp-server/rtsp-media.h:
13478 media: allow configuration of allowed lower transport
13480 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13482 * gst/rtsp-server/rtsp-client.h:
13483 * gst/rtsp-server/rtsp-media.c:
13484 * gst/rtsp-server/rtsp-media.h:
13485 * gst/rtsp-server/rtsp-sdp.c:
13486 * gst/rtsp-server/rtsp-sdp.h:
13487 * gst/rtsp-server/rtsp-server.c:
13488 rtsp: keep track of server ip and ipv6
13489 Keep track of how the client connected to the server and setup the udp ports
13490 with the same protocol.
13491 Copy the server ip address in the SDP so that clients can send RTCP back to
13494 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13496 * gst/rtsp-server/rtsp-session.c:
13499 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13501 * gst/rtsp-server/rtsp-client.c:
13502 client: use right size for malloc
13504 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13506 * gst/rtsp-server/rtsp-server.c:
13507 server: comment ipv6 server listening address
13509 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13511 * gst/rtsp-server/rtsp-media.c:
13512 media: allow for ipv6 sockets
13514 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13516 * gst/rtsp-server/rtsp-server.c:
13517 * gst/rtsp-server/rtsp-server.h:
13518 server: rework server part
13519 Allow setting a bind address, make sure we can deal with ipv6.
13520 Remove the port property and change with the service property.
13522 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13524 * gst/rtsp-server/rtsp-media.h:
13525 media: update comments a little
13527 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13529 * gst/rtsp-server/rtsp-client.c:
13530 client: make content-base better
13531 Use the URI formatting functions to make a content-base. Also make sure that
13532 there is a trailing / at the end.
13534 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13536 * gst/rtsp-server/rtsp-client.c:
13537 client: guard against invalid paths
13539 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13541 * examples/test-video.c:
13542 test: catch server bind errors
13544 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13546 * gst/rtsp-server/rtsp-media.c:
13547 rtspmedia: emit "unprepared" if _prepare fails.
13548 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13549 media object is removed from its factory's cache.
13551 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13553 * gst/rtsp-server/rtsp-media.c:
13554 media: collect media position when seek completes
13556 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13558 * gst/rtsp-server/rtsp-client.c:
13559 client: call unlink_streams in client finalize
13562 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13564 * gst/rtsp-server/rtsp-media.c:
13565 media: limit the time to wait to something huge
13566 Avoid waiting forever but limit the timeout to 20 seconds.
13568 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13570 * gst/rtsp-server/rtsp-sdp.c:
13571 sdp: reindent and check for prepared status
13573 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13575 * gst/rtsp-server/rtsp-media.c:
13576 * gst/rtsp-server/rtsp-media.h:
13577 * gst/rtsp-server/rtsp-session.c:
13578 media: avoid doing _get_state() for state changes
13579 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13580 until the media is prerolled or in error. This avoids doing a blocking call of
13581 gst_element_get_state() that can cause lockups when there is an error.
13584 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13586 * gst/rtsp-server/rtsp-media.c:
13589 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13591 * gst/rtsp-server/rtsp-media-factory.c:
13592 media-factory: better error handling
13593 Improve the error handling a bit.
13595 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13597 * gst/rtsp-server/rtsp-client.c:
13598 client: rework transport parsing
13599 Rework the transport parsing code so that we can ignore transports we don't
13600 support instead of just picking the first one we can parse.
13601 Configure a (for now hardcoded) destination for multicast transports.
13603 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13605 * gst/rtsp-server/rtsp-media.c:
13606 media: set multicast sink parameters
13607 Disable loop and automatic multicast join on the udpsink elements.
13608 Add some more debug info.
13609 Reset some state variables in the right place.
13610 Use the right port numbers for multicast.
13612 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13614 * gst/rtsp-server/rtsp-session.c:
13615 session: handle transport setup correctly
13616 Handle UDP, MCAST and TCP transport negotiation more correctly.
13617 Store the server session SSRC in the transport.
13619 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13621 * gst/rtsp-server/rtsp-client.c:
13622 rtsp-client: implement error_full
13623 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13626 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13629 * gst/rtsp-server/rtsp-client.c:
13630 * gst/rtsp-server/rtsp-server.c:
13631 docs: update docs and comments
13633 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13635 * gst/rtsp-server/rtsp-sdp.c:
13636 sdp: make server work better when behind a proxy
13638 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13640 * gst/rtsp-server/rtsp-client.c:
13641 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13643 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13645 * gst/rtsp-server/rtsp-client.c:
13646 * gst/rtsp-server/rtsp-media-factory.c:
13647 * gst/rtsp-server/rtsp-media-mapping.c:
13648 * gst/rtsp-server/rtsp-media.c:
13649 * gst/rtsp-server/rtsp-server.c:
13650 * gst/rtsp-server/rtsp-session-pool.c:
13651 * gst/rtsp-server/rtsp-session.c:
13652 Use GStreamer's debugging subsystem
13654 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13656 * gst/rtsp-server/rtsp-media-factory.c:
13657 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13659 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13662 back to development
13664 === release 0.10.5 ===
13666 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13671 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13674 configure: bump required versions
13676 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13678 * gst/rtsp-server/rtsp-client.c:
13679 client: call weak-unref on client->sessions from finalize
13682 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13684 * gst/rtsp-server/rtsp-media.c:
13685 media: Fixed crasher where caps got unref'ed too often
13687 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13690 * pkgconfig/.gitignore:
13691 * pkgconfig/Makefile.am:
13692 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13693 Added pkg-config file to use gst-rtsp-server uninstalled
13695 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13697 * gst/rtsp-server/rtsp-media.c:
13698 media: add some docs
13700 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13702 * gst/rtsp-server/rtsp-client.c:
13703 rtsp: Use gst_rtsp_watch_send_message().
13704 Use gst_rtsp_watch_send_message() since the old API which used
13705 gst_rtsp_watch_queue_message() has been deprecated.
13707 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13710 back to development
13712 === release 0.10.4 ===
13714 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13719 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13721 * gst/rtsp-server/rtsp-client.c:
13722 * gst/rtsp-server/rtsp-session.c:
13723 * gst/rtsp-server/rtsp-session.h:
13724 rtsp: allocate channels in TCP mode
13725 When the client does not provide us with channels in TCP mode, allocate channels
13728 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13730 * gst/rtsp-server/rtsp-client.c:
13731 client: don't crash when tunnelid is missing
13732 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13733 don't crash but return an error response to the client.
13736 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13738 * bindings/vala/gst-rtsp-server-0.10.vapi:
13739 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13740 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13741 bindings: update vala bindings with new method
13743 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13745 * gst/rtsp-server/rtsp-session-pool.c:
13746 * gst/rtsp-server/rtsp-session-pool.h:
13747 sessionpool: add function to filter sessions
13748 Add generic function to retrieve/remove sessions.
13750 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13753 configure: bump core/base requirements to release
13755 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13757 * gst/rtsp-server/rtsp-media.c:
13758 media: fix indentation
13760 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13762 * gst/rtsp-server/rtsp-media.c:
13763 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13765 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13767 * gst/rtsp-server/rtsp-media.c:
13768 set state and remove elements of media in for loop
13770 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13772 * bindings/vala/gst-rtsp-server-0.10.vapi:
13773 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13774 Added gst_rtsp_media_remove_elements function to Vala bindings
13776 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13778 * gst/rtsp-server/rtsp-media.c:
13779 * gst/rtsp-server/rtsp-media.h:
13780 Added gst_rtsp_media_remove_elements function
13782 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13784 * gst/rtsp-server/rtsp-media.c:
13785 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13787 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13789 * bindings/vala/gst-rtsp-server-0.10.vapi:
13790 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13791 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13792 Updated Vala bindings
13794 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13796 * gst/rtsp-server/rtsp-media.c:
13797 * gst/rtsp-server/rtsp-media.h:
13798 Added vmethod unprepare to GstRTSPMedia
13799 The default implementation sets the state of the pipeline to GST_STATE_NULL
13801 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13803 * gst/rtsp-server/rtsp-media-factory.c:
13804 * gst/rtsp-server/rtsp-media-factory.h:
13805 Made collect_streams function public
13807 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13809 * gst/rtsp-server/rtsp-media-factory.c:
13810 * gst/rtsp-server/rtsp-media-factory.h:
13811 * gst/rtsp-server/rtsp-media.c:
13812 Added vmethod create_pipeline to GstRTSPMediaFactory
13813 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13815 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13817 * gst/rtsp-server/rtsp-client.c:
13818 client: use g_source_destroy()
13819 We need to use g_source_destroy() because we might have added the source to a
13820 different main context than the default one.
13822 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13824 * gst/rtsp-server/Makefile.am:
13825 * gst/rtsp-server/rtsp-client.c:
13826 * gst/rtsp-server/rtsp-params.c:
13827 * gst/rtsp-server/rtsp-params.h:
13828 rtsp: prepare for handling GET/SET_PARAMETER
13829 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13831 Fix return codes of handlers.
13833 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13835 * gst/rtsp-server/rtsp-media.c:
13836 media: don't leak session pads
13838 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13840 * gst/rtsp-server/rtsp-media.c:
13841 media: clean up the messages a bit
13843 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13845 * gst/rtsp-server/rtsp-sdp.c:
13846 sdp: warn and skip streams without media
13848 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13850 * bindings/vala/gst-rtsp-server-0.10.vapi:
13851 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13852 vala: Fixed typo in header file of RTSPMediaStream
13854 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13856 * gst/rtsp-server/rtsp-media.c:
13858 Fix a debug message
13859 Make dumping RTCP stats configurable
13861 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13863 * gst/rtsp-server/rtsp-media.c:
13864 media: be less verbose and leak less
13866 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13868 * gst/rtsp-server/rtsp-media.c:
13869 media: don't leak the destination address
13871 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13873 * gst/rtsp-server/rtsp-client.c:
13874 * gst/rtsp-server/rtsp-media.c:
13875 * gst/rtsp-server/rtsp-media.h:
13876 * gst/rtsp-server/rtsp-session.c:
13877 * gst/rtsp-server/rtsp-session.h:
13878 rtsp: use RTCP to keep the session alive
13879 Use the RTCP rtcp-from stats field to find the associated session and use this
13880 to keep the session alive.
13882 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13884 * gst/rtsp-server/rtsp-session.c:
13885 session: add 5sec to the real session timeout
13886 Allow the session to live 5sec longer before really timing out. This should give
13887 clients some extra time to keep the session active.
13889 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13891 * gst/rtsp-server/rtsp-client.c:
13892 client: replay OK to GET/SET_PARAMETER
13893 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13894 so that we return OK for those requests.
13896 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13898 * gst/rtsp-server/rtsp-media.c:
13899 * gst/rtsp-server/rtsp-media.h:
13900 media: keep track of active transports
13901 Keep track of which transport is active to avoid closing the connection too
13903 Remove the destination transport also when going to NULL.
13904 Print some stats about the SDES and other RTCP messages we receive from the
13907 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13909 * examples/.gitignore:
13910 * examples/Makefile.am:
13911 * examples/test-sdp.c:
13912 example: add SDP relay example
13914 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13916 * gst/rtsp-server/rtsp-media.c:
13917 media: also count active TCP connections
13919 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13921 * gst/rtsp-server/rtsp-media-factory.c:
13922 * gst/rtsp-server/rtsp-media.c:
13923 * gst/rtsp-server/rtsp-media.h:
13924 rtsp: add support for dynamic elements
13925 Add support for dynamic elements.
13926 Don't set live pipelines back to paused.
13928 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13930 * gst/rtsp-server/rtsp-sdp.c:
13931 sdp: don't add encoding name when absent in caps
13933 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13935 * gst/rtsp-server/rtsp-client.c:
13936 client: warn when we can't do RTP-Info
13938 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13940 * gst/rtsp-server/rtsp-media-factory.c:
13941 factory: factor out the stream construction
13943 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13945 * gst/rtsp-server/rtsp-client.c:
13946 client: only add RTP-Info when we have the info
13947 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13950 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13953 back to development
13955 === release 0.10.3 ===
13957 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13961 - Fixes a bug where it put the wrong verion in pkgconfig
13962 - Link RTP and RTCP sources
13964 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13966 * gst/rtsp-server/rtsp-media.c:
13967 * gst/rtsp-server/rtsp-media.h:
13968 media: link the RTP udpsrc to the session manager
13969 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13970 shut down when the client sends a packet to open firewalls.
13972 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13974 * pkgconfig/gst-rtsp-server.pc.in:
13975 Don't use hard-coded version number in pkg-config file
13977 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13980 back to development
13982 === release 0.10.2 ===
13984 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13989 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13992 * common/m4/.gitignore:
13993 * examples/.gitignore:
13994 * pkgconfig/.gitignore:
13995 add some .gitignore files
13997 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13999 * gst/rtsp-server/rtsp-media.c:
14000 media: seek to key frames
14002 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14004 * gst/rtsp-server/rtsp-media.c:
14005 media: emit the unprepared signal by id
14006 Emit the unprepared signal by id instead of name and set the media as
14009 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14011 * gst/rtsp-server/rtsp-media.c:
14012 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
14014 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14016 * gst/rtsp-server/rtsp-server.c:
14017 Added finalize function to GstRTPSPServer to unref session pool and media mapping
14019 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14021 * bindings/vala/gst-rtsp-server-0.10.vapi:
14022 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14023 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14024 Updated vala bindings
14026 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14028 * gst/rtsp-server/Makefile.am:
14029 * gst/rtsp-server/rtsp-client.c:
14030 * gst/rtsp-server/rtsp-media.c:
14031 server: use appsink and appsrc with the API
14032 Use the appsink/appsrc API instead of the signals for higher
14035 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14037 * examples/test-ogg.c:
14038 tests: set the payload type correctly
14040 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14042 * gst/rtsp-server/rtsp-media-factory.c:
14043 factory: connect to the unprepare signal
14044 Connect to the unprepare signal for non-reusable media so that we can remove
14045 them from the cache.
14047 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14049 * gst/rtsp-server/rtsp-media.c:
14050 * gst/rtsp-server/rtsp-media.h:
14051 media: add signal to notify of unprepare
14053 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14055 * gst/rtsp-server/rtsp-media.c:
14056 * gst/rtsp-server/rtsp-media.h:
14057 media: more work on making the media shared
14058 Add a reusable flag to medias, indicating that they can be reused after a state
14062 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14064 * examples/test-readme.c:
14065 examples: mark the example as shared for testing
14067 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14069 * gst/rtsp-server/rtsp-media.c:
14070 * gst/rtsp-server/rtsp-media.h:
14071 client: support shared media
14072 Always perform the state actions even if the target state of the pipeline is
14073 already correct, we still want to add/remove the transports when we are dealing
14075 Keep a counter of the number of active transports for a media so that we can use
14076 this to perform a state change when needed.
14077 Perform a state change of the pipeline only when the first transport was added
14078 or when there are no active transports.
14080 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14082 * gst/rtsp-server/rtsp-client.c:
14083 client: fix refcounting crasher
14084 Don't need to remove the weak refs in the finalize methods, they are already
14085 removed in the dispose.
14086 Don't register the callback with a DestroyNofity.
14088 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14090 * gst/rtsp-server/rtsp-client.c:
14091 Fix rtsp client refcount management in TCP mode.
14092 Don't unref a client ref we never had. Fixes an unref
14093 of an already-free client object after a client
14094 teardown request for me.
14096 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14098 * gst/rtsp-server/rtsp-session.c:
14099 docs: fix typo in API docs
14101 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14103 * gst/rtsp-server/rtsp-media.c:
14104 More seeking fixes.
14105 Keep the udp sources in playing even if we go to paused. unlock the sources when
14107 Add some more debug info.
14108 Only seek when we need to.
14109 Keep track of the position when we go to paused.
14111 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14113 * gst/rtsp-server/rtsp-client.c:
14114 * gst/rtsp-server/rtsp-media.c:
14115 * gst/rtsp-server/rtsp-media.h:
14116 Add beginnings of seeking.
14117 Parse the Range header and perform a seek on the pipeline for the requested
14118 position. It's disabled currently until I figure out what's going wrong.
14120 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14122 * gst/rtsp-server/rtsp-client.c:
14123 allow pause requests for now.
14126 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14128 * gst/rtsp-server/rtsp-client.c:
14129 Remove weak ref on the session in teardown
14130 We need to remove our weakref from the session when we do a teardown because
14131 else we close the TCP connection prematurely.
14133 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14135 * gst/rtsp-server/rtsp-client.c:
14136 * gst/rtsp-server/rtsp-client.h:
14137 * gst/rtsp-server/rtsp-session-pool.c:
14138 Do some more session cleanup
14139 Make session timeout kill the TCP connection that currently watches the
14141 Remove the client timeout property.
14143 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14145 * gst/rtsp-server/rtsp-client.c:
14146 * gst/rtsp-server/rtsp-client.h:
14147 * gst/rtsp-server/rtsp-media.c:
14148 * gst/rtsp-server/rtsp-media.h:
14149 * gst/rtsp-server/rtsp-server.c:
14150 * gst/rtsp-server/rtsp-session.c:
14151 * gst/rtsp-server/rtsp-session.h:
14153 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
14156 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14158 * examples/Makefile.am:
14159 * examples/test-launch.c:
14160 Add example server that takes launch lines
14161 Add an example server that streams any -launch line.
14163 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14165 * examples/test-readme.c:
14166 * gst/rtsp-server/rtsp-client.c:
14167 * gst/rtsp-server/rtsp-media.c:
14168 * gst/rtsp-server/rtsp-media.h:
14169 Add support for live streams
14170 Add support for live streams and ranges
14171 Start on handling TCP data transfer.
14173 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14175 * gst/rtsp-server/rtsp-media.c:
14176 Free the pipeline before other things
14179 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14181 * gst/rtsp-server/rtsp-client.c:
14182 Only free the pending tunnel if there is one
14185 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14187 * gst/rtsp-server/rtsp-client.c:
14188 * gst/rtsp-server/rtsp-client.h:
14189 * gst/rtsp-server/rtsp-media.c:
14190 rtsp-server: Add support for tunneling
14191 Add support for tunneling over HTTP.
14192 Use new connection methods to retrieve the url.
14193 Dispatch messages based on the message type instead of blindly
14194 assuming it's always a request.
14195 Keep track of the watch id so that we can remove it later.
14196 Set the media pipeline to NULL before unreffing the pipeline.
14198 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14200 * gst/rtsp-server/rtsp-client.c:
14201 * gst/rtsp-server/rtsp-client.h:
14202 Fix for channel -> watch rename in gstreamer
14203 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
14205 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14207 * gst/rtsp-server/rtsp-client.c:
14208 * gst/rtsp-server/rtsp-client.h:
14210 Use the async RTSP channels instead of spawning a new thread for each client.
14211 If a sessionid is specified in a request, fail if we don't have the session.
14213 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14215 * gst/rtsp-server/rtsp-media.c:
14216 Add better debug info
14217 Add some better debug info.
14219 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14221 * examples/test-video.c:
14223 Add support for session timeouts in the example.
14225 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14227 * gst/rtsp-server/rtsp-session-pool.c:
14228 * gst/rtsp-server/rtsp-session-pool.h:
14229 Pass GTimeVal around for performance reasons
14230 Get the current time only once and pass it around so that sessions don't have to
14231 get the current time anymore.
14232 Add experimental support for a GSource that dispatches when the session needs to
14235 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14237 * gst/rtsp-server/rtsp-session.c:
14238 * gst/rtsp-server/rtsp-session.h:
14239 Add better support for session timeouts
14240 Add a method to request the number of milliseconds when a session will timeout.
14242 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14244 * gst/rtsp-server/rtsp-media.c:
14245 * gst/rtsp-server/rtsp-media.h:
14246 Add suport for RTP manager monitoring
14247 Add the first stage in monitoring the rtp manager.
14248 Make sure we don't update the state to something we don't want.
14250 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14252 * gst/rtsp-server/rtsp-client.c:
14253 Add support for session keepalive
14254 Get and update the session timeout for all requests. get the session as early as
14257 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14259 * gst/rtsp-server/rtsp-media-factory.h:
14260 * gst/rtsp-server/rtsp-media.c:
14261 * gst/rtsp-server/rtsp-media.h:
14262 Handle media bus messages
14263 Handle media bus messages in a custom mainloop and dispatch them to the
14264 RTSPMedia objects. Let the default implementation handle some common messages.
14266 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14268 * gst/rtsp-server/rtsp-client.c:
14269 * gst/rtsp-server/rtsp-session-pool.c:
14270 * gst/rtsp-server/rtsp-session.c:
14271 Some more session timeout handling
14272 Move the session header setting code to a central place so that we always add
14273 the timeout parameter too.
14274 Handle timeouts by running the session cleanup code.
14275 Stop media before cleaning up.
14277 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14279 * gst/rtsp-server/rtsp-client.c:
14280 * gst/rtsp-server/rtsp-client.h:
14281 Add timeout property
14282 Add a timeout property ot the client and make the other properties into GObject
14285 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14287 * gst/rtsp-server/rtsp-session-pool.c:
14288 Use getters and setters in property code
14289 Use the getters and setters for the timeout property instead of locking
14292 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14294 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
14296 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14298 * gst/rtsp-server/rtsp-session-pool.c:
14299 * gst/rtsp-server/rtsp-session-pool.h:
14300 * gst/rtsp-server/rtsp-session.c:
14301 * gst/rtsp-server/rtsp-session.h:
14302 Add more timeout stuff
14303 Add method to check if a session is expired.
14304 Add method to perform cleanup on a session pool.
14306 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14308 * gst/rtsp-server/rtsp-client.c:
14309 * gst/rtsp-server/rtsp-session-pool.c:
14310 * gst/rtsp-server/rtsp-session-pool.h:
14311 * gst/rtsp-server/rtsp-session.c:
14312 * gst/rtsp-server/rtsp-session.h:
14313 Add beginnings of session timeouts and limits
14314 Add the timeout value to the Session header for unusual timeout values.
14315 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14316 limit on the amount of retry we do after a sessionid collision.
14317 Add properties to the sessionid and the timeout of a session. Keep track of
14318 creation time and last access time for sessions.
14320 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14322 * gst/rtsp-server/rtsp-client.c:
14323 * gst/rtsp-server/rtsp-media.c:
14324 * gst/rtsp-server/rtsp-media.h:
14325 * gst/rtsp-server/rtsp-sdp.c:
14326 * gst/rtsp-server/rtsp-session-pool.c:
14327 * gst/rtsp-server/rtsp-session.c:
14328 * gst/rtsp-server/rtsp-session.h:
14329 Cleanup of sessions and more
14330 Fix the refcounting of media and sessions in the client. Properly clean up the
14331 session data when the client performs a teardown.
14332 Add Server header to responses.
14333 Allow for multiple uri setups in one session.
14334 Add Range header to the PLAY response and add the range attribute to the SDP
14336 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14337 give the ownership of the sessionid to the session object.
14339 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14341 * gst/rtsp-server/rtsp-server.c:
14342 * gst/rtsp-server/rtsp-server.h:
14344 Rename the 'server_port' variable to simply 'port'.
14346 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14349 * gst/rtsp-server/rtsp-client.c:
14350 * gst/rtsp-server/rtsp-media.c:
14351 * gst/rtsp-server/rtsp-media.h:
14352 * gst/rtsp-server/rtsp-session.c:
14353 * gst/rtsp-server/rtsp-session.h:
14354 Rework the way we handle transports for streams
14355 Make the media accept an array of transports for the streams that we have
14356 configured for the play/pause requests.
14357 Implement server states for a client and its media.
14358 Require 0.10.22.1 (git HEAD) of gstreamer.
14360 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14362 * gst/rtsp-server/rtsp-client.c:
14363 * gst/rtsp-server/rtsp-media-factory.c:
14364 Drop const from functions dealing with urls
14365 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14366 have the right const in them.
14368 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14370 * gst/rtsp-server/rtsp-client.c:
14371 * gst/rtsp-server/rtsp-media.c:
14372 * gst/rtsp-server/rtsp-sdp.c:
14376 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14378 * gst/rtsp-server/rtsp-client.c:
14379 * gst/rtsp-server/rtsp-media-factory.c:
14380 * gst/rtsp-server/rtsp-media.c:
14381 * gst/rtsp-server/rtsp-media.h:
14383 Don't keep a reference to the GstRTSPMedia in the stream.
14384 Free more things when freeing the GstRTSPMedia.
14386 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14389 * gst/rtsp-server/rtsp-media-factory.c:
14390 * gst/rtsp-server/rtsp-media-factory.h:
14391 * gst/rtsp-server/rtsp-media.c:
14392 * gst/rtsp-server/rtsp-media.h:
14393 * gst/rtsp-server/rtsp-server.c:
14394 * gst/rtsp-server/rtsp-server.h:
14395 More docs and small cleanups
14396 Add some more docs and update the README
14397 Cleanup some method names.
14398 Remove an unneeded idx field in the GstRTSPMediaStream
14400 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14403 * examples/Makefile.am:
14404 * examples/test-readme.c:
14405 Add a README and more example code
14406 Add a README file that contains a small introduction on how to use the server
14407 along with the example code explained in the readme.
14409 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14411 * gst/rtsp-server/rtsp-media.c:
14412 * gst/rtsp-server/rtsp-server.c:
14413 Fix some leaks and change default port
14414 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14415 we finished the initial preroll. If we keep them locked, setting the pipeline to
14416 NULL will not stop and clean up the sources correctly.
14417 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14419 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14421 * gst/rtsp-server/rtsp-session.c:
14422 * gst/rtsp-server/rtsp-session.h:
14423 Cleanups to the session object
14424 Remove some unneeded variables in the session state of a stream such as the
14425 owner media and the server transport.
14426 Get the configuration of a media stream in a session based on the media_stream
14427 in the original object instead of our cached index.
14428 Free more data in the finalize method.
14430 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14432 * gst/rtsp-server/rtsp-client.c:
14433 * gst/rtsp-server/rtsp-client.h:
14434 Cleanups and reuse media from DESCRIBE
14435 Handle thread create errors.
14436 Rename some internal methods to better match what they actually do.
14437 Handle misconfiguration of session_pool and media_mapping gracefully.
14438 Cache the DESCRIBE media and uri in the client connection and reuse them when
14439 we receive a SETUP request in the same connection for the same uri.
14440 Cleanup the client connection object.
14442 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14444 * gst/rtsp-server/rtsp-media-factory.c:
14445 * gst/rtsp-server/rtsp-media-factory.h:
14446 * gst/rtsp-server/rtsp-media.c:
14447 * gst/rtsp-server/rtsp-media.h:
14448 Add shared properties to media and factory
14449 Add the shared property to media.
14450 Implement some simple caching in the factory depending on if the media is shared
14453 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14455 * gst/rtsp-server/rtsp-client.c:
14456 Add a little comment
14457 Add some comment about the content-base header.
14459 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14461 * examples/Makefile.am:
14462 * examples/test-mp4.c:
14463 * examples/test-ogg.c:
14464 * examples/test-video.c:
14465 * gst/rtsp-server/Makefile.am:
14466 * gst/rtsp-server/rtsp-client.c:
14467 * gst/rtsp-server/rtsp-client.h:
14468 * gst/rtsp-server/rtsp-media-factory.c:
14469 * gst/rtsp-server/rtsp-media-factory.h:
14470 * gst/rtsp-server/rtsp-media.c:
14471 * gst/rtsp-server/rtsp-media.h:
14472 * gst/rtsp-server/rtsp-sdp.c:
14473 * gst/rtsp-server/rtsp-sdp.h:
14474 * gst/rtsp-server/rtsp-server.c:
14475 * gst/rtsp-server/rtsp-server.h:
14476 * gst/rtsp-server/rtsp-session.c:
14477 * gst/rtsp-server/rtsp-session.h:
14478 Reorganize things, prepare for media sharing
14479 Added various other test server examples
14480 Move the SDP message generation to a separate helper.
14481 Refactor common code for finding the session.
14482 Add content-base for realplayer compatibility
14483 Clean up request uris before processing for better vlc compatibility.
14484 Move prerolling and pipeline construction to the RTSPMedia object.
14485 Use multiudpsink for future pipeline reuse.
14487 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14490 Back to development
14493 === release 0.10.1 ===
14495 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14498 Make 0.10.1 release
14501 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14503 * bindings/vala/Makefile.am:
14505 Add more directories and files to the dist.
14507 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14509 * bindings/python/Makefile.am:
14510 * bindings/python/rtspserver.override:
14511 Fixed compile error of python bindings
14513 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14515 * bindings/vala/gst-rtsp-server-0.10.vapi:
14516 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14517 Marked values as nullable accordingly
14519 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14521 * bindings/vala/gst-rtsp-server-0.10.vapi:
14522 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14523 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14524 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14525 Updated Vala bindings
14527 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14529 * gst/rtsp-server/rtsp-client.c:
14530 * gst/rtsp-server/rtsp-media-mapping.c:
14531 * gst/rtsp-server/rtsp-media-mapping.h:
14532 * gst/rtsp-server/rtsp-media.h:
14533 * gst/rtsp-server/rtsp-session-pool.h:
14534 Cleanups and doc updates
14535 Add some more documentation and do some minor cleanups here and there.
14537 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14539 * gst/rtsp-server/rtsp-client.c:
14540 * gst/rtsp-server/rtsp-media-factory.c:
14541 * gst/rtsp-server/rtsp-media-factory.h:
14542 * gst/rtsp-server/rtsp-media.c:
14543 * gst/rtsp-server/rtsp-media.h:
14544 * gst/rtsp-server/rtsp-session.c:
14545 * gst/rtsp-server/rtsp-session.h:
14547 Rename GstRTSPMediaBin to GstRTSPMedia
14548 Parse the request url into a GstRTSPUri object and pass this object to the
14549 various handlers and methods that require the uri.
14551 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14555 Add some more docs and remove some old code from the example.
14557 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14559 * gst/rtsp-server/rtsp-client.c:
14560 Handle state change failures better
14561 Handle state change failures better when changing the state of the pipeline to
14564 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14566 * gst/rtsp-server/rtsp-media-factory.c:
14567 * gst/rtsp-server/rtsp-media-factory.h:
14568 Make element creation more extendible
14569 Add get_element vmethod to the default MediaFactory so that subclasses can just
14570 override that method and still use the default logic for making a MediaBin from
14573 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14576 * gst/rtsp-server/Makefile.am:
14577 * gst/rtsp-server/rtsp-client.c:
14578 * gst/rtsp-server/rtsp-client.h:
14579 * gst/rtsp-server/rtsp-media-factory.c:
14580 * gst/rtsp-server/rtsp-media-factory.h:
14581 * gst/rtsp-server/rtsp-media-mapping.c:
14582 * gst/rtsp-server/rtsp-media-mapping.h:
14583 * gst/rtsp-server/rtsp-media.c:
14584 * gst/rtsp-server/rtsp-media.h:
14585 * gst/rtsp-server/rtsp-server.c:
14586 * gst/rtsp-server/rtsp-server.h:
14587 * gst/rtsp-server/rtsp-session.c:
14588 * gst/rtsp-server/rtsp-session.h:
14589 Make the server handle arbitrary pipelines
14590 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14591 The GstMediaBin object has a handle to a bin with elements and to a list of
14592 GstMediaStream objects that this bin produces.
14593 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14594 with methods to register and remove those mappings.
14595 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14596 used by the server instance.
14597 Modify the example application so that it shows how to create custom pipelines
14598 attached to a specific mount point.
14599 Various misc cleanps.
14601 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14603 * gst/rtsp-server/rtsp-server.c:
14604 * gst/rtsp-server/rtsp-server.h:
14605 Allow setting a custom media factory for a server
14607 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14609 * gst/rtsp-server/rtsp-client.c:
14610 * gst/rtsp-server/rtsp-client.h:
14611 Allow setting a custom media factory for a client.
14613 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14615 * gst/rtsp-server/Makefile.am:
14616 Add Makefile entry for the media factory
14618 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14620 * gst/rtsp-server/rtsp-media-factory.c:
14621 * gst/rtsp-server/rtsp-media-factory.h:
14622 Add media factory to map urls to media pipeline objects.
14624 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14626 * gst/rtsp-server/rtsp-media.c:
14627 * gst/rtsp-server/rtsp-media.h:
14628 Add comments. Remove unused field
14630 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14632 * gst/rtsp-server/rtsp-session-pool.c:
14633 * gst/rtsp-server/rtsp-session-pool.h:
14634 Allow custom session pools to override the session id allocation algorithms Add some comments.
14636 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14638 * gst/rtsp-server/rtsp-session.h:
14641 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14643 * gst/rtsp-server/rtsp-client.c:
14644 * gst/rtsp-server/rtsp-client.h:
14645 Move the connection code in one place Add some comments
14647 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14649 * gst/rtsp-server/rtsp-server.c:
14650 * gst/rtsp-server/rtsp-server.h:
14651 Make vmethod to create and accept new clients. Add some docs.
14653 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14655 * gst/rtsp-server/rtsp-server.c:
14656 * gst/rtsp-server/rtsp-server.h:
14657 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14659 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14661 * gst/rtsp-server/rtsp-client.c:
14662 * gst/rtsp-server/rtsp-client.h:
14663 Name the parameters more appropriately.
14665 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14667 * gst/rtsp-server/rtsp-session-pool.c:
14668 Do some more cleanup of the session pool.
14670 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14672 * gst/rtsp-server/Makefile.am:
14673 * gst/rtsp-server/rtsp-client.c:
14674 Check if return value of gst_rtsp_session_get_media is not NULL
14676 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14678 * gst/rtsp-server/Makefile.am:
14679 Install rtsp-session and rtsp-session-pool headers
14681 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14686 * bindings/python/Makefile.am:
14687 * bindings/python/arg-types.py:
14688 * bindings/python/codegen/Makefile.am:
14689 * bindings/python/codegen/__init__.py:
14690 * bindings/python/codegen/argtypes.py:
14691 * bindings/python/codegen/code-coverage.py:
14692 * bindings/python/codegen/codegen.py:
14693 * bindings/python/codegen/definitions.py:
14694 * bindings/python/codegen/defsparser.py:
14695 * bindings/python/codegen/docextract.py:
14696 * bindings/python/codegen/docgen.py:
14697 * bindings/python/codegen/fileprefix.override:
14698 * bindings/python/codegen/fileprefixmodule.c:
14699 * bindings/python/codegen/h2def.py:
14700 * bindings/python/codegen/mergedefs.py:
14701 * bindings/python/codegen/mkskel.py:
14702 * bindings/python/codegen/override.py:
14703 * bindings/python/codegen/reversewrapper.py:
14704 * bindings/python/codegen/scmexpr.py:
14705 * bindings/python/rtspserver-types.defs:
14706 * bindings/python/rtspserver.defs:
14707 * bindings/python/rtspserver.override:
14708 * bindings/python/rtspservermodule.c:
14710 Add python bindings.
14712 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14714 * bindings/Makefile.am:
14716 Don't go into python dir when requirements for python bindings are missing
14718 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14720 * bindings/Makefile.am:
14721 * bindings/vala/Makefile.am:
14723 Install Vala bindings if vala is available
14725 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14727 * bindings/vala/gst-rtsp-server-0.10.deps:
14728 * bindings/vala/gst-rtsp-server-0.10.vapi:
14729 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14730 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14731 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14732 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14733 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14734 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14735 Regenerated Vala bindings
14737 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14739 * bindings/vala/gst-rtsp-server.vapi:
14740 * bindings/vala/packages/gst-rtsp-server.metadata:
14741 Fixed typo in included headers for vala bindings
14743 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14747 * pkgconfig/Makefile.am:
14748 * pkgconfig/gst-rtsp-server.pc.in:
14749 Added pkgconfig file
14751 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14753 * bindings/vala/gst-rtsp-server.vapi:
14754 * bindings/vala/packages/gst-rtsp-server.excludes:
14755 * bindings/vala/packages/gst-rtsp-server.gi:
14756 * bindings/vala/packages/gst-rtsp-server.metadata:
14757 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14759 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14761 * bindings/vala/gst-rtsp-server.vapi:
14762 * bindings/vala/packages/gst-rtsp-server.deps:
14763 * bindings/vala/packages/gst-rtsp-server.files:
14764 * bindings/vala/packages/gst-rtsp-server.gi:
14765 * bindings/vala/packages/gst-rtsp-server.metadata:
14766 * bindings/vala/packages/gst-rtsp-server.namespace:
14767 Added Vala bindings
14769 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14771 * gst/rtsp-server/rtsp-session.c:
14772 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14774 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14776 * examples/Makefile.am:
14777 * gst/rtsp-server/Makefile.am:
14778 Put GStreamer version in library name
14780 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14782 * examples/Makefile.am:
14783 * gst/rtsp-server/Makefile.am:
14784 Fix some issues to pass distcheck
14786 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14788 * gst/rtsp-server/rtsp-server.c:
14789 Added port property to GstRTSPServer class.
14791 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14796 * examples/Makefile.am:
14799 * gst/rtsp-server/Makefile.am:
14800 * gst/rtsp-server/rtsp-client.c:
14801 * gst/rtsp-server/rtsp-client.h:
14802 * gst/rtsp-server/rtsp-media.c:
14803 * gst/rtsp-server/rtsp-media.h:
14804 * gst/rtsp-server/rtsp-server.c:
14805 * gst/rtsp-server/rtsp-server.h:
14806 * gst/rtsp-server/rtsp-session-pool.c:
14807 * gst/rtsp-server/rtsp-session-pool.h:
14808 * gst/rtsp-server/rtsp-session.c:
14809 * gst/rtsp-server/rtsp-session.h:
14811 Split in library and example program
14813 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14815 * src/rtsp-client.h:
14816 Removed obsolete variable
14818 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14820 * src/rtsp-client.c:
14821 * src/rtsp-client.h:
14822 Removed pipeline variable GstRTSPClient, because it's only used in one function
14824 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14826 * src/rtsp-media.c:
14827 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14829 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14831 * src/rtsp-session.c:
14832 Initialize some more vars.
14834 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14836 * src/rtsp-session.c:
14837 Initialize variable to avoid compiler warning.
14839 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14842 Add a reasonable generic .gitignore