3 2022-11-07 23:53:59 +0000 Tim-Philipp Müller <tim@centricular.com>
7 * docs/plugins/gst_plugins_cache.json:
8 * gst-rtsp-server.doap:
12 2022-11-07 23:53:57 +0000 Tim-Philipp Müller <tim@centricular.com>
15 Update ChangeLogs for 1.21.2
17 2022-10-25 09:39:07 +0300 Sebastian Dröge <sebastian@centricular.com>
19 * gst/rtsp-server/rtsp-server.c:
20 Fix various warnings from gobject-introspection
21 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3261>
23 2022-10-15 12:40:04 +0300 Sebastian Dröge <sebastian@centricular.com>
25 * gst/rtsp-server/rtsp-auth.c:
26 * gst/rtsp-server/rtsp-latency-bin.c:
27 * gst/rtsp-server/rtsp-media-factory.c:
28 * gst/rtsp-server/rtsp-media.c:
29 * gst/rtsp-server/rtsp-onvif-media-factory.c:
30 * gst/rtsp-server/rtsp-server.c:
31 * gst/rtsp-server/rtsp-stream.c:
32 rtsp-server: Add/fix various annotations
33 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
35 2022-10-14 08:53:18 +0200 Edward Hervey <edward@centricular.com>
37 * gst/rtsp-server/rtsp-client.h:
38 rtsp-client: Remove duplicate documentation
39 Confuses the documentation builder, since it's documented twice it complains
40 about a missing "Since:" marker whereas it's present in the documentation
42 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
44 2022-08-19 16:16:26 +0200 Linus Svensson <linussn@axis.com>
46 * gst/rtsp-server/rtsp-server.c:
47 rtsp-server: Free client if no connection could be created
48 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>
50 2022-10-11 14:55:48 +0200 Peter Stensson <petest@axis.com>
52 * gst/rtsp-server/rtsp-client.h:
53 rtsp-server: Add since marker for adjust_error_code
54 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>
56 2022-06-21 09:51:55 +0200 Peter Stensson <petest@axis.com>
58 * gst/rtsp-server/rtsp-client.c:
59 * gst/rtsp-server/rtsp-client.h:
60 * gst/rtsp-server/rtsp-media.c:
61 * tests/check/gst/client.c:
62 * tests/check/gst/media.c:
63 rtsp-server: Add support for adjusting request response on pipeline errors
64 The idea is to give the application the possibility to adjust the error
65 code when responding to a request. For that purpose the pipeline's bus
66 messages are emitted to subscribers through a signal handle-message.
67 The subscribers can then check those messages for errors and adjust
68 the response error code by overriding the virtual method
71 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
73 2022-10-04 03:57:31 +0100 Tim-Philipp Müller <tim@centricular.com>
75 * docs/plugins/gst_plugins_cache.json:
78 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
80 === release 1.21.1 ===
82 2022-10-04 01:14:01 +0100 Tim-Philipp Müller <tim@centricular.com>
87 * docs/plugins/gst_plugins_cache.json:
88 * gst-rtsp-server.doap:
92 2022-10-04 01:13:59 +0100 Tim-Philipp Müller <tim@centricular.com>
95 Update ChangeLogs for 1.21.1
97 2022-09-21 19:19:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
100 meson: Use implicit builtin dirs in pkgconfig generation
101 Starting with Meson 0.62, meson automatically populates the variables
102 list in the pkgconfig file if you reference builtin directories in the
103 pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
104 We need this, because ${prefix}/libexec is a hard-coded value which is
105 incorrect on, for example, Debian.
106 Bump requirement to 0.62, and remove version compares that retained
107 support for older Meson versions.
108 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
109 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
111 2021-03-24 14:20:18 -0500 Zebediah Figura <z.figura12@gmail.com>
114 meson: Build with -Wl,-z,nodelete to prevent unloading of dynamic libraries and plugins
115 GLib made the unfortunate decision to prevent libgobject from ever being
116 unloaded, which means that now any library which registers a static type
117 can't ever be unloaded either (and any library that depends on those,
119 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
121 2022-09-05 13:28:18 +1200 Chris Wiggins <chris@chriswiggins.co.nz>
123 * gst/rtsp-server/rtsp-context.c:
124 * gst/rtsp-server/rtsp-context.h:
125 rtsp-server: context: Add method to set the RTSPToken on some RTSPContext
127 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2979>
129 2022-08-24 19:50:19 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
131 * gst/rtsp-server/rtsp-server-internal.h:
132 * gst/rtsp-server/rtsp-stream-transport.c:
133 * gst/rtsp-server/rtsp-stream.c:
134 gst-rtsp-server: Fix pushing backlog to client
135 Check back pressure of a stream transport before popping buffer from its backlog.
136 If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
138 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
140 2022-09-02 16:31:54 +0300 Sebastian Dröge <sebastian@centricular.com>
142 * gst/rtsp-server/rtsp-stream.c:
143 rtsp-server: stream: Don't loop forever if binding to the multicast address fails
144 The address/port is pre-defined by the caller of the function, so
145 retrying is only going to loop forever.
146 Ideally the multicast address should be checked after allocating but
147 this doesn't happen currently, so it's better to error out cleanly then
148 to loop forever trying the same address.
149 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
151 2022-09-01 15:11:31 -0400 Thibault Saunier <tsaunier@igalia.com>
153 * gst/rtsp-sink/meson.build:
155 meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
156 Removing some copy pasted code
157 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
159 2022-09-01 11:51:48 -0400 Thibault Saunier <tsaunier@igalia.com>
162 * gst/rtsp-server/meson.build:
164 meson: Namespace the plugins_doc_dep/libraries variables
165 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
167 2022-08-31 18:44:14 -0400 Thibault Saunier <tsaunier@igalia.com>
170 meson: Rename plugins list and make them "dependency" objects
171 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
173 2022-05-25 18:40:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
175 * gst/rtsp-sink/gstrtspclientsink.c:
176 rtsp+rtmp: Forward warning added to tls-validation-flags to our users
177 With the 2.72 release, glib-networking developers have decided that
178 TLS certificate validation cannot be implemented correctly by them, so
179 they've deprecated it.
180 In a nutshell: a cert can have several validation errors, but there
181 are no guarantees that the TLS backend will return all those errors,
182 and things are made even more complicated by the fact that the list of
183 errors might refer to certs that are added for backwards-compat and
184 won't actually be used by the TLS library.
185 Our best option is to ignore the deprecation and pass the warning onto
186 users so they can make an appropriate security decision regarding
188 We can't deprecate the tls-validation-flags property because it is
189 very useful when connecting to RTSP cameras that will never get
190 updates to fix certificate errors.
191 Relevant upstream merge requests / issues:
192 https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214
193 https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179
194 https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
195 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
197 2022-07-12 16:58:00 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
199 * gst/rtsp-server/rtsp-client.c:
200 rtsp-client: Fix url for generating key in media factory
201 The mount point at / can be accessed by both the URL forms rtsp://<IP>:<PORT> and rtsp://<IP>:<PORT>/.
202 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.
203 This commit creates a new GstRTSPUrl as the normalized url to send to gst_rtsp_media_factory_construct().
204 Fixes:https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1297
205 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2681>
207 2022-06-29 10:55:13 +0100 Tim-Philipp Müller <tim@centricular.com>
210 coding style: allow declarations after statement
211 See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
212 and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
213 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2683>
215 2022-06-14 16:18:35 +0100 Tim-Philipp Müller <tim@centricular.com>
218 * docs/plugins/gst_plugins_cache.json:
219 * docs/plugins/index.md:
220 * docs/plugins/sitemap.txt:
221 docs: make sure rtspclientsink plugin docs index page is called index.html
222 .. instead of plugin-index.html.
223 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2592>
225 2022-04-06 12:56:30 +0100 Tim-Philipp Müller <tim@centricular.com>
228 Bump GLib requirement to >= 2.62
229 Can't require 2.64 yet because of
230 https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323
231 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
233 2022-05-16 18:06:16 +0200 Patricia Muscalu <patricia@axis.com>
235 * gst/rtsp-server/rtsp-media.c:
236 rtsp-media: Correct logic on GstRTSPStreamBlocking message reception
237 We must take into account the receiving streams as well when calculating
238 the expected number of the received GstRTSPStreamBlocking messages.
239 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2429>
241 2022-04-27 01:13:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
243 * tests/check/gst/onvif.c:
244 tests/onvif: improve robustness
245 The previous iteration of the code was inferring the type of the
246 frame by looking at the overall size of the gst-payloaded packet.
247 It is more robust to actually parse the payload and look at the
248 actual data buffers it contains.
249 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
251 2022-04-27 01:10:46 +0200 Mathieu Duponchelle <mathieu@centricular.com>
253 * tests/check/gst/onvif.c:
254 tests/onvif: don't push buffers outside segment
255 segment->stop is exclusive, so in reverse playback mode we do not
256 need to output a buffer at that position as it will simply get
258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
260 2022-02-15 13:39:43 +0000 Pierre Bourré <pierre.moltess@gmail.com>
262 * gst/rtsp-sink/gstrtspclientsink.c:
263 rtspclientsink: fix possible shutdown deadlock collect_streams()
264 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1696>
266 2022-04-07 19:14:27 +0300 Sebastian Dröge <sebastian@centricular.com>
268 * gst/rtsp-server/rtsp-sdp.c:
269 rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME
270 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
272 2022-04-13 14:34:57 +0200 Marc Leeman <m.leeman@televic.com>
274 * gst/rtsp-server/rtsp-stream.c:
275 gst-rtsp-server: minor spelling fixes
276 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>
278 2022-03-25 15:00:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
280 * examples/meson.build:
282 Remove glib and gobject dependencies everywhere
283 They are part of gst_dep already and we have to make sure to always have
284 gst_dep. The order in dependencies matters, because it is also the order
285 in which Meson will set -I args. We want gstreamer's config.h to take
286 precedence over glib's private config.h when it's a subproject.
287 While at it, remove useless fallback args for gmodule/gio dependencies,
288 only gstreamer core needs it.
289 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
291 2022-03-28 21:03:16 +1100 Matthew Waters <matthew@centricular.com>
293 * gst/rtsp-server/rtsp-stream.c:
294 rtsp-stream: remove unused variable:
296 ../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
297 guint n_messages = 0;
299 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
301 2022-03-18 13:42:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
304 meson: Bump all meson requirements to 0.60
305 Lots of new warnings ever since
306 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1934
307 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
309 2022-02-23 17:39:18 +0100 Vivienne Watermeier <vwatermeier@igalia.com>
311 * gst/rtsp-server/rtsp-token.c:
312 documentation: improve misleading wording
313 The documentation for several gst_*_writable_structure functions stated
314 that they would never return NULL, without making clear that the passed
315 object is required to be writable. This changes the wording in those
316 cases to make that requirement more clear.
317 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
319 2022-02-10 08:01:02 +0100 Branko Subasic <branko@axis.com>
321 * examples/test-onvif-server.c:
322 * tests/check/gst/onvif.c:
323 rtponviftimestamp: add support for using reference timestamps
324 Make it posible to configure the element to obtain the timestamps from
325 reference timestamp meta data instead of using the ntp-offset property,
326 or estimating its own offset. Currently the only time format supported
327 is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.
328 In addition the custom event GstNtpOffset has been renamed to
329 GstOnvifTimestamp, to reflect that it is not necessarily used to convey
330 the ntp-offset. As a consequence we had to modify a couple of files in
331 the rtsp-server as well.
333 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
335 2022-02-18 16:05:53 +0100 Branko Subasic <branko@axis.com>
337 * tests/check/gst/onvif.c:
338 * tests/check/gst/rtspserver.c:
339 * tests/check/gst/stream.c:
340 gst-rtsp-server: Plug a few memory leaks in tests
341 Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and
342 gst_stream tests by running the tests in valgrind.
343 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
345 2022-03-07 09:14:46 +0100 Branko Subasic <branko@axis.com>
347 * gst/rtsp-server/rtsp-client.c:
348 gst-rtsp-server: fix race in rtsp-client
349 When tunneling over HTTP, if connection on the second channel happens
350 before the control timer is created we may trigger an assert in
351 rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
352 attaching the client thread to the context.
354 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
356 2022-02-04 11:15:47 +0000 Tim-Philipp Müller <tim@centricular.com>
358 * docs/gst_plugins_cache.json:
361 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
363 === release 1.20.0 ===
365 2022-02-03 19:53:25 +0000 Tim-Philipp Müller <tim@centricular.com>
370 * docs/gst_plugins_cache.json:
371 * gst-rtsp-server.doap:
375 2022-02-03 19:53:18 +0000 Tim-Philipp Müller <tim@centricular.com>
378 Update ChangeLogs for 1.20.0
380 === release 1.19.90 ===
382 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
387 * docs/gst_plugins_cache.json:
388 * gst-rtsp-server.doap:
392 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
395 Update ChangeLogs for 1.19.90
397 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
399 * examples/test-appsrc2.c:
400 gst-rtsp-server: Fix leak in appsrc2 example
401 In the need-data appsrc callback, a buffer is pulled from the
402 appsink. This buffer is then copied so that metadata is writable.
403 The copy is pushed to the appsrc but it doesn't take ownership
404 of the buffer so we need to manually unref it. The original buffer
405 is finally unreffed when the sample is freed.
406 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
408 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
412 meson: Add explicit check: kwarg to all run_command() calls
413 This is required since Meson 0.61.0, and causes a warning to be
415 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
416 https://github.com/mesonbuild/meson/issues/9300
417 This exposed a bunch of places where we had broken run_command()
418 calls, unnecessary run_command() calls, and places where check: true
420 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
422 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
424 * gst/rtsp-server/meson.build:
425 rtsp-server: add gst_dep to gst_rtsp_server_deps
426 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
427 will avoid the following build failure, because the correct girdir
428 location will be retrieved from gstreamer-1.0.pc:
429 /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
430 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
431 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
432 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
433 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
435 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
436 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
438 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
440 * gst/rtsp-server/rtsp-stream.c:
441 rtsp-stream: fix get_rates raciness
442 Prior to this patch, we considered that a stream was blocking
443 whenever a pad probe was triggered for either the RTP pad or
445 This led to situations where we subsequently unblocked and expected
446 to find a segment on the RTP pad, which was racy.
447 Instead, we now only consider that the stream is blocking when
448 the pad probe for the RTP pad has triggered with a blockable object
449 (buffer, buffer list, gap event).
450 The RTCP pad is simply blocked without affecting the state of the
453 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
455 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
457 * docs/gst_plugins_cache.json:
461 === release 1.19.3 ===
463 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
468 * docs/gst_plugins_cache.json:
469 * gst-rtsp-server.doap:
473 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
476 Update ChangeLogs for 1.19.3
478 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
481 meson: require matching GStreamer dep versions for unstable development releases
482 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
483 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
485 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
487 * tests/check/meson.build:
488 meson: update for meson.build_root() and .build_source() deprecation
489 -> use meson.project_build_root() or .global_build_root() instead.
490 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
492 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
495 * tests/check/meson.build:
496 meson: update for dep.get_pkgconfig_variable() deprecation
497 ... in favour of dep.get_variable('foo', ..) which in some
498 cases allows for further cleanups in future since we can
499 extract variables from pkg-config dependencies as well as
500 internal dependencies using this mechanism.
501 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
503 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
505 * gst/rtsp-server/meson.build:
506 * gst/rtsp-sink/meson.build:
507 rtsp-server: define G_LOG_DOMAIN
509 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
511 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
514 meson: bump meson requirement to >= 0.59
515 For monorepo build and ugly/bad, for advanced feature
516 option API like get_option('xyz').required(..) which
517 we use in combination with the 'gpl' option.
518 For rest of modules for consistency (people will likely
519 use newer features based on the top-level requirement).
520 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
522 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
525 meson: Streamline the way we detect when to build documentation
526 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
528 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
531 * gst/rtsp-server/meson.build:
533 meson: List libraries and their corresponding gir definition
534 Introduces a `libraries` variable that contains all libraries in a
535 list with the following format:
539 'lib': library_object
540 'gir': [ {full gir definition in a dict } ]
545 It therefore refactors the way we build the gir so that we can reuse the
546 same information to build them against 'gstreamer-full' in gst-build
547 when linking statically
548 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
550 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
552 * gst/rtsp-server/meson.build:
553 meson: Mark files as files()
554 Making it more robust and future proof
555 And fix issues that it creates
556 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
558 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
560 * gst/rtsp-server/rtsp-media.c:
561 rtsp-media: Unprepare suspended medias too
562 Previously suspended medias immediately reached the UNPREPARED state
563 without going through the media's unprepare() vfunc. This didn't allow
564 the media subclass to do any additional cleanup, and for example the
565 shutdown-eos property of GstRTSPMedia was ignored.
566 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
568 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
570 * gst/rtsp-server/rtsp-media.c:
571 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
572 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
574 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
576 * gst/rtsp-server/rtsp-client.c:
577 * gst/rtsp-server/rtsp-session.c:
578 * gst/rtsp-server/rtsp-session.h:
579 rtsp-client: make sure sessmedia will not get freed while used
580 handle_*_request() functions were all retrieving the session media from
581 the session by calling gst_rtsp_session_get_media () which is a transfer-none
582 call. If a session timeout happens at that time, the session media may get freed
583 making the pointer invalid..
585 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
587 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
589 * gst/rtsp-server/rtsp-media.c:
590 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
591 Previously the status was only changed for other medias.
592 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
594 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
596 * gst/rtsp-server/rtsp-session.c:
597 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
598 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
599 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
601 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
604 doc: update IRC links to OFTC
605 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
607 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
609 * docs/gst_plugins_cache.json:
612 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
614 === release 1.19.2 ===
616 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
621 * docs/gst_plugins_cache.json:
622 * gst-rtsp-server.doap:
626 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
628 * gst/rtsp-server/rtsp-media.c:
629 * gst/rtsp-server/rtsp-stream.c:
630 * gst/rtsp-server/rtsp-stream.h:
631 * gst/rtsp-sink/gstrtspclientsink.c:
632 Protection against early RTCP packets.
633 When receiving RTCP packets early the funnel is not ready yet and
634 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
635 This causes the thread that handle RTCP packets to go to pause mode.
636 Since this thread is in pause mode there will be no further callbacks to
637 handle keep-alive for incoming RTCP packets. This will make the session
638 time out if the client is not using another keep-alive mechanism.
639 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
640 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
642 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
646 Update COPYING.LIB, COPYING files
647 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
649 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
651 * docs/gst_plugins_cache.json:
655 === release 1.19.1 ===
657 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
662 * docs/gst_plugins_cache.json:
663 * gst-rtsp-server.doap:
667 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
669 * gst/rtsp-server/rtsp-stream.c:
670 rtsp-stream: use new gst_buffer_new_memdup()
671 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
673 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
675 * gst/rtsp-server/rtsp-media-factory-uri.c:
676 rtsp-media: fix leak when adding converter
677 Free the previous caps before reusing the variable for the converter caps.
678 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
680 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
682 * gst/rtsp-server/rtsp-client.c:
683 rtsp-client: fix leak adding headers
684 gst_rtsp_message_add_header() makes a copy of the header, instead
686 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
688 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
690 * gst/rtsp-server/rtsp-stream.c:
691 Use gst_element_request_pad_simple...
692 Instead of the deprecated gst_element_get_request_pad.
693 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
695 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
697 * gst/rtsp-server/rtsp-media.c:
698 rtsp-media: Ensure the bus watch is removed during unprepare
699 It's possible for the destruction of the source to be delayed.
700 Instead of relying on the dispose() to remove the bus watch, do
702 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
704 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
707 docs: minor spelling correction in README
708 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
710 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
712 * examples/test-replay-server.c:
713 test-replay-server: minor spelling corrections
714 Bumped on these while investigating the example code.
715 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
717 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
719 * tests/check/gst/stream.c:
720 tests: Don't fail tests if IPv6 not available.
721 On computers with IPv6 disabled it shouldn't result in a test failure.
722 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
724 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
726 * gst/rtsp-server/rtsp-media.c:
727 rtsp-media: Add one more case to seek avoidance
728 This is an extension to the previous commit. There can also be cases where the
729 start position is not specified, in those cases we should also avoid doing
730 seeking unless it's forced.
731 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
733 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
735 * gst/rtsp-server/rtsp-media.c:
736 rtsp-media: Improve skipping trickmode seek.
737 We can also skip the seek if the end range is already
739 Avoids initial seek on play start if playing full stream.
740 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
742 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
744 * gst/rtsp-sink/gstrtspclientsink.c:
745 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
746 It's sufficient to run them during the FIRST stage instead of in both.
747 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
749 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
751 * tests/check/gst/rtspclientsink.c:
752 tests: rtspclientsink: fix some leaks
753 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
755 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
757 * gst/rtsp-sink/gstrtspclientsink.c:
758 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
759 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
761 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
763 * tests/check/gst/rtspclientsink.c:
764 rtspclientsink: add unit test for potential shutdown deadlock
765 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
767 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
769 * gst/rtsp-sink/gstrtspclientsink.c:
770 rtspclientsink: fix deadlock on shutdown before preroll
771 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
772 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
774 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
776 * gst/rtsp-server/rtsp-stream.c:
777 rtsp-stream: avoid deadlock in send_func
778 Currently the send_func() runs in a thread of its own which is started
779 the first time we enter handle_new_sample(). It runs in an outer loop
780 until priv->continue_sending is FALSE, which happens when a TEARDOWN
781 request is received. We use a local variable, cont, which is initialized
782 to TRUE, meaning that we will always enter the outer loop, and at the
783 end of the outer loop we assign it the value of priv->continue_sending.
784 Within the outer loop there is an inner loop, where we wait to be
785 signaled when there is more data to send. The inner loop is exited when
786 priv->send_cookie has changed value, which it does when more data is
787 available or when a TEARDOWN has been received.
788 But if we get a TEARDOWN before send_func() is entered we will get stuck
789 in the inner loop because no one will increase priv->session_cookie
791 By not entering the outer loop in send_func() if priv->continue_sending
792 is FALSE we make sure that we do not get stuck in send_func()'s inner
793 loop should we receive a TEARDOWN before the send thread has started.
794 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
795 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
797 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
799 * gst/rtsp-server/rtsp-client.c:
800 rtsp-client: cleanup transports during TEARDOWN
801 When tunneling RTP over RTSP the stream transports are stored in a hash
802 table in the GstRTSPClientPrivate struct. They are used for, among other
803 things, mapping channel id to stream transports when receiving data from
804 the client. The stream tranports are created and added to the hash table
805 in handle_setup_request(), but unfortuately they are not removed in
806 handle_teardown_request(). This means that if the client sends data on
807 the RTSP connection after it has sent the TEARDOWN, which is often the
808 case when audio backchannel is enabled, handle_data() will still be able
809 to map the channel to a session transport and pass the data along to it.
810 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
811 because the stream is no longer joined to a bin.
812 We avoid this by removing the stream transports from the hash table when
813 we handle the TEARDOWN request.
814 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
816 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
818 * docs/gst_plugins_cache.json:
819 * gst/rtsp-sink/gstrtspclientsink.c:
820 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
821 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
823 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
825 * tests/check/gst/client.c:
826 Add test cases for mountpoint of '/'
827 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
829 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
831 * gst/rtsp-server/rtsp-client.c:
832 * gst/rtsp-server/rtsp-mount-points.c:
833 * gst/rtsp-server/rtsp-session-media.c:
834 Make a mount point of "/" work correctly.
835 As far as I can tell, this is neither explicitly allowed nor
836 forbidden by RFC 7826.
837 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
838 use in the wild (presumably with non-GStreamer servers).
839 GStreamer's prior behavior was confusing, in that
840 gst_rtsp_mount_points_add_factory() would appear to accept a mount
841 path of "" or "/", but later connection attempts would fail with a
842 "media not found" error.
843 This commit makes a mount path of "/" work for either form of URL,
844 while an empty mount path ("") is rejected and logs a warning.
845 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
847 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
849 * docs/gst_plugins_cache.json:
850 * gst/rtsp-sink/gstrtspclientsink.c:
851 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
852 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
854 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
856 * gst/rtsp-server/rtsp-media.c:
857 rtsp-media: Only count senders when counting blocked streams
858 Only sender streams sends the GstRTSPStreamBlocking message, so only
859 these should be counted before setting media status to prepared.
860 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
862 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
864 * gst/rtsp-sink/gstrtspclientsink.c:
865 rtspclientsink add proper support for uri queries
866 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
868 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
870 * gst/rtsp-server/rtsp-client.c:
871 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
872 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
873 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
875 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
877 * gst/rtsp-server/rtsp-stream.c:
878 rtsp-stream: collect a clock_rate when blocking
879 This lets us provide a clock_rate in a fashion similar to the
880 other code paths in get_rtpinfo()
881 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
883 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
885 * gst/rtsp-server/rtsp-media.c:
886 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
887 Otherwise this will cause memory corruption as the property expects a 64
889 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
891 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
893 * gst/rtsp-server/rtsp-media.c:
894 * gst/rtsp-server/rtsp-stream.c:
895 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
896 To prevent cases with prerolling when the inactive stream prerolls first
897 and the server proceeds without waiting for the active stream, we will
898 ignore GstRTSPStreamBlocking messages from incomplete streams. When
899 there are no complete streams (during DESCRIBE), we will listen to all
901 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
903 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
905 * tests/check/gst/media.c:
906 * tests/check/meson.build:
907 * tests/files/test.avi:
908 media test: Add test for seeking one active stream with a demuxer
909 Add another seek_one_active_stream test but with a demuxer. The demuxer
910 will flush both streams in opposed to the existing test which only
911 flushes the active stream. This will help exposing problems with the
912 prerolling process after a flushing seek.
913 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
915 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
917 * gst/rtsp-server/meson.build:
919 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
920 * pkgconfig/gstreamer-rtsp-server.pc.in:
921 * pkgconfig/meson.build:
922 Meson: Use pkg-config generator
923 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
925 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
928 meson: update glib minimum version to 2.56
929 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
931 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
933 * examples/test-launch.c:
934 * gst/rtsp-server/rtsp-media-factory.c:
935 * gst/rtsp-server/rtsp-media-factory.h:
936 * gst/rtsp-server/rtsp-media.c:
937 * gst/rtsp-server/rtsp-server-internal.h:
938 * gst/rtsp-server/rtsp-stream.c:
939 * tests/check/gst/client.c:
940 rtsp-media-factory: expose API to disable RTCP
941 This is supported by the RFC, and can be useful on systems where
942 allocating two consecutive ports is problematic, and RTCP is not
944 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
946 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
948 * hooks/pre-commit.hook:
950 git: use our standard pre commit hook
951 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
953 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
955 * gst/rtsp-server/rtsp-stream.c:
956 rtsp-stream: make use of blocked_running_time in query_position
957 When blocking, the sink element will not have received a buffer
958 yet and the position query will fail. Instead, we make use of
959 the running time of the buffer we blocked on.
960 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
962 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
964 * gst/rtsp-server/rtsp-stream.c:
965 rtsp-stream: collect rtp info when blocking
966 We don't unblock the stream anymore before replying to the
967 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
968 so the sinks don't have a last-sample after potentially flush
969 seeking. seek_trickmode waits for preroll however, which means
970 the stream will block and wait for a first buffer. Subsequent
971 calls to get_rtpinfo() can thus make use of the information.
972 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
973 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
975 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
977 * examples/meson.build:
978 * examples/test-replay-server.c:
979 * examples/test-replay-server.h:
980 examples: Add an example for loop playback
981 This demo example shows a way of file loop playback of a given source.
982 Note that client seek request is not properly implemented yet.
983 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
985 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
987 * gst/rtsp-server/rtsp-media.c:
988 rtsp-media: Plug memory leak
989 The get-storage signal of rtpbin increases the ref count of the storage.
990 So we have to unref it after usage.
991 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
993 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
995 * gst/rtsp-server/rtsp-media.c:
996 rtsp-media: Get rates only on sender streams
997 When play a media with both sender and receiver stream, like ONVIF
998 back channel audio in, gst_rtsp_media_get_rates call
999 gst_rtsp_stream_get_rates for each stream to set the rates. But
1000 gst_rtsp_stream_get_rates return false for the receiver steam, which
1001 lead a g_assert crash.
1002 Instead to get rates on all streams, now just get rates on sender
1004 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
1006 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1008 * gst/rtsp-server/rtsp-media.c:
1009 * gst/rtsp-server/rtsp-server-internal.h:
1010 * gst/rtsp-server/rtsp-stream.c:
1011 rtsp-media: set a 0 storage size for TCP receivers
1012 ulpfec correction is obviously useless when receiving a stream
1013 over TCP, and in TCP modes the rtp storage receives non
1014 timestamped buffers, causing it to queue buffers indefinitely,
1015 until the queue grows so large that sanity checks kick in and
1016 warnings start to get emitted.
1017 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
1019 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1021 * gst/rtsp-server/rtsp-stream.c:
1022 rtsp-stream: preroll on gap events
1023 This allows negotiating a SDP with all streams present, but only
1024 start sending packets at some later point in time
1025 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
1027 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1029 * gst/rtsp-server/rtsp-media.c:
1030 rtsp-media: do not unblock on unsuspend
1031 rtsp_media_unsuspend() is called from handle_play_request()
1032 before sending the play response. Unblocking the streams here
1033 was causing data to be sent out before the client was ready
1034 to handle it, with obvious side effects such as initial packets
1035 getting discarded, causing decoding errors.
1036 Instead we can simply let the media streams be unblocked when
1037 the state of the media is set to PLAYING, which occurs after
1038 sending the play response.
1039 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
1041 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1044 ci: include template from gst-ci master branch again
1046 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
1048 * docs/gst_plugins_cache.json:
1052 === release 1.18.0 ===
1054 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1060 * docs/gst_plugins_cache.json:
1061 * gst-rtsp-server.doap:
1065 === release 1.17.90 ===
1067 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1072 * docs/gst_plugins_cache.json:
1073 * gst-rtsp-server.doap:
1077 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
1079 * gst/rtsp-server/rtsp-thread-pool.c:
1080 rtsp-thread-pool.c: fix clang 10 warning
1081 clang 10 is complaining about incompatible types due to the
1084 ../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]
1086 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1088 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
1090 * gst/rtsp-server/rtsp-thread-pool.c:
1091 rtsp-thread-pool.c: fix clang 10 warning
1092 clang 10 is complaining about incompatible types due to the
1095 ../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]
1097 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1099 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
1101 * gst/rtsp-server/rtsp-sdp.c:
1102 rtsp-sdp: Fix resource leak in mikey messsage
1103 Fixed a resource leak for mikey message while adding crypto session
1105 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
1107 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
1110 * scripts/extract-release-date-from-doap-file.py:
1111 meson: set release date from .doap file for releases
1112 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
1114 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1116 * gst/rtsp-server/rtsp-stream.c:
1117 rtsp-stream: explicitly set caps on udpsrc elements
1118 This causes them to send caps events before data flow, which is
1119 usually a pretty correct thing to do!
1120 Not doing so manifested in a bug where ssrcdemux wouldn't forward
1121 the caps it had received with an extra ssrc field, as it hadn't
1122 received any caps event.
1124 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
1126 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1128 * docs/gst_plugins_cache.json:
1132 === release 1.17.2 ===
1134 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1139 * docs/gst_plugins_cache.json:
1140 * gst-rtsp-server.doap:
1144 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
1146 * docs/gst_plugins_cache.json:
1147 doc: Stop documenting properties from parents
1149 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1151 * docs/gst_plugins_cache.json:
1152 docs: Fix version in the plugins cache
1153 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1155 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1157 * gst/rtsp-sink/gstrtspclientsink.c:
1158 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
1159 It's deprecated, unneeded and doesn't do anything anymore.
1160 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1162 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
1167 === release 1.17.1 ===
1169 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
1174 * docs/gst_plugins_cache.json:
1175 * gst-rtsp-server.doap:
1179 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1181 * gst/rtsp-server/rtsp-media.c:
1182 rtsp-media: Add/configure transports when completing the pipeline
1183 Otherwise the transports are not set up yet during the PLAY request
1184 handling when unsuspending (and thus unblocking) the media.
1185 In case of live pipelines this then causes the first few packets to go
1186 to the sinks before they know what to do with them, and they simply
1187 discard them which is rather suboptimal in case of keyframes.
1188 For non-live pipelines this is not a problem because the sink will still
1189 be PAUSED and as such not send out the data yet but wait until it goes
1190 to PLAYING, which is late enough.
1191 Adding the transports multiple times is not a problem: if the transport
1192 is already added it won't be added another time and TRUE will be
1194 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
1196 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
1197 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1199 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1201 * gst/rtsp-server/rtsp-media.c:
1202 rtsp-media: Fix misleading comment
1203 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1205 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
1207 * gst/rtsp-server/rtsp-media.c:
1208 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
1209 The pad probes are not needed anymore at this point and later when
1210 reaching buffering 100% only the state is changed, no unblocking
1212 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1214 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
1216 * gst/rtsp-server/rtsp-media.c:
1217 rtsp-media: Remove duplicated media_unblock() function
1218 It does literally the same as media_streams_set_blocked(FALSE).
1219 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1221 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
1223 * examples/test-onvif-server.c:
1224 test-onvif-server: cast ntp-offset property value to 64 bit
1225 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
1227 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
1229 * docs/gst_plugins_cache.json:
1230 docs: Update plugins cache
1232 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1234 * examples/test-onvif-server.c:
1235 * examples/test-onvif-server.h:
1236 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1237 onvif-media-factory: define autoptr cleanup function
1238 And have the factory in the onvif-server example inherit from
1239 GstRTSPOnvifMediaFactory.
1240 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
1242 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
1244 * docs/gst_plugins_cache.json:
1245 docs: Update plugins cache
1247 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
1249 * tests/check/gst/rtspserver.c:
1250 tests: enforce I420 format
1251 Test was not enforcing a video format on videotestsrc. I420 was picked as it
1252 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
1253 true (gst-plugins-base!689).
1254 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
1256 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1258 * gst/rtsp-sink/gstrtspclientsink.c:
1259 plugins: uddate gst_type_mark_as_plugin_api() calls
1261 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
1264 doc: Require hotdoc >= 0.11.0
1266 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1268 * docs/gst_plugins_cache.json:
1269 docs: Update gst_plugins_cache.json
1271 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1273 * gst/rtsp-sink/gstrtspclientsink.c:
1274 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
1276 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1278 * gst/rtsp-server/meson.build:
1279 meson: gir: remove bogus sources_top_dir kwarg
1280 Doesn't actually exist. Was fixed differently in Meson
1281 so that the user doesn't have to specify it.
1282 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
1284 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1286 * tests/check/meson.build:
1287 tests: put registry into tests/check not the gst/ subdir
1288 Underscorify the test name before setting GST_REGISTRY,
1289 so the registry actually ends up in the current build dir
1290 and not some subdir.
1291 For consistency with the other modules, but should also
1292 avoid problems on windows.
1293 Also fix indentation of environment block.
1294 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1296 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
1298 * tests/check/meson.build:
1299 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
1300 If core is built as a subproject (e.g. as in gst-build), make sure to use
1301 the gst-plugin-scanner from the built subproject. Without this, gstreamer
1302 might accidentally use the gst-plugin-scanner from the install prefix if
1303 that exists, which in turn might drag in gst library versions we didn't
1304 mean to drag in. Those gst library versions might then be older than
1305 what our current build needs, and might cause our newly-built plugins
1306 to get blacklisted in the test registry because they rely on a symbol
1307 that the wrongly-pulled in gst lib doesn't have.
1308 This should fix running of unit tests in gst-build when invoking
1309 meson test or ninja test from outside the devenv for the case where
1310 there is an older or different-version gst-plugin-scanner installed
1311 in the install prefix.
1312 In case no gst-plugin-scanner is installed in the install prefix, this
1313 will fix "GStreamer-WARNING: External plugin loader failed. This most
1314 likely means that the plugin loader helper binary was not found or
1315 could not be run. You might need to set the GST_PLUGIN_SCANNER
1316 environment variable if your setup is unusual." warnings when running
1318 In the case where we find GStreamer core via pkg-config we use
1319 a newly-added pkg-config var "pluginscannerdir" to get the right
1320 directory. This has the benefit of working transparently for both
1321 installed and uninstalled pkg-config files/setups.
1322 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1324 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
1326 * tests/check/meson.build:
1327 tests: gst-plugins-base and -bad plugins are required for the unit tests
1328 Make hard requirement until we have more fine-grained control
1329 in the unit tests. Of course the presence of the .pc file doesn't
1330 imply that the plugins we need are actually there, but it's at
1331 least a step in the right direction.
1332 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1334 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1336 * tests/check/meson.build:
1337 tests: pick up rtsp-server plugins from build directory only
1338 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1340 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
1342 * gst/rtsp-server/rtsp-media.c:
1343 rtsp-media: wait for all GstRTSPStreamBlocking messages
1344 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
1345 each active stream when checking if all streams are blocked.
1346 Without this change there will be a race condition when using two or
1347 more streams and rtsp-media receives a GstRTSPStreamBlocking message
1348 from one of the streams. This is because rtsp-media then checks if all
1349 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
1350 stream. This function call returns TRUE if the stream has sent a
1351 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
1352 receive this message. This would then result in that rtsp-media
1353 erroneously thinks it is blocking all streams which could result in
1354 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
1355 preroll, this could result in that rtsp-media thinks that the pipeline
1356 is prerolled even though that might not be the case.
1357 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
1359 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
1361 * gst/rtsp-server/rtsp-media.c:
1362 rtsp-media: update expected_async_done during suspend
1363 Set expected_async_done to FALSE in default_suspend() if a state change
1364 occurs and the return value from set_target_state() is something other
1365 than GST_STATE_CHANGE_ASYNC.
1366 Without this change there is a risk that expected_async_done will be
1367 TRUE even though no asynchronous state change is taking place. This
1368 could happen if the pipeline is set to PAUSED using
1369 media_set_pipeline_state_locked(), an asynchronous state change starts
1370 and then the media is suspended (which could result in a state change,
1371 aborting the asynchronous state change). If the media is suspended
1372 before the asynchronous state change ends then expected_async_done will
1373 be TRUE but no asynchronous state change is taking place.
1374 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
1376 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
1378 * gst/rtsp-server/rtsp-client.c:
1379 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1380 There was a race condition where client was being finalized and
1381 concurrently in some other thread the rtsp ctrl timout was relying on
1382 client data that was being freed.
1383 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1384 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1386 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1388 * gst/rtsp-server/rtsp-media-factory.c:
1389 * gst/rtsp-server/rtsp-media-factory.h:
1390 * gst/rtsp-server/rtsp-media.c:
1391 * gst/rtsp-server/rtsp-media.h:
1392 media-factory: complete DSCP QoS setting support
1393 add dscp_qos setting support at factory and media level to setup IP DSCP
1394 field of bounded UDP sinks.
1395 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1396 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1398 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1400 * gst/rtsp-server/rtsp-client.c:
1401 rtsp-client: Fix some race conditions around timeout source removal
1402 We always need to take the lock while accessing it as otherwise another
1403 thread might've removed it in the meantime. Also when destroying and
1404 creating a new one, ensure that the mutex is not shortly unlocked in
1405 between as during that time another one might potentially be created
1407 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1409 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1411 * gst/rtsp-server/rtsp-media.c:
1412 * gst/rtsp-server/rtsp-stream.c:
1413 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1414 And the same for gst_rtsp_stream_get_rates().
1415 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1417 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1419 * examples/test-onvif-server.c:
1420 examples: test-onvif-server: fix compiler warnings on raspbian
1421 Fix printf format for 64-bit variables.
1422 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1424 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1426 * gst/rtsp-server/rtsp-stream-transport.c:
1427 * gst/rtsp-server/rtsp-stream-transport.h:
1428 * gst/rtsp-server/rtsp-stream.c:
1429 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1430 The old API is preserved now and new API was added that provides the
1431 additional parameter to the callback.
1432 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1433 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1435 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1437 * gst/rtsp-server/rtsp-client.c:
1438 rtsp-client: Store the timeout source by pointer instead of id
1439 That way we don't have to retrieve it again from the main context when
1440 destroying it but can directly do so.
1441 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1443 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1445 * gst/rtsp-server/rtsp-client.c:
1446 rtsp-client: Clean up watch/watch context and related state consistently
1447 And assert that it was cleaned up properly before the client is
1448 finalized. If something is still around when the client is shut down
1449 then something went very wrong before.
1450 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1452 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1454 * gst/rtsp-server/rtsp-client.c:
1455 * tests/check/gst/rtspserver.c:
1456 rtsp-client: Combine the pre-session and post-session timeout
1457 They previously used the same state but different mechanisms and
1458 functions, which was difficult to follow, error prone and simply
1460 Also adjust the test for the post-session timeout a bit to be less racy
1461 now that the timing has slightly changed.
1462 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1464 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1466 * gst/rtsp-server/rtsp-client.c:
1467 rtsp-client: Don't ever close the client connection directly when a session is torn down
1468 There might be other sessions that are running over the same RTSP
1469 connection and we should not simply close the client directly if one of
1471 By default the connection will be closed once the client closes it or
1472 the OS does. This behaviour can be adjusted with the
1473 post-session-timeout property, which allows to close it automatically
1474 from the server side after all sessions are gone and the given timeout
1476 This reverts the previous commit.
1477 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1479 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1481 * gst/rtsp-server/rtsp-client.c:
1482 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1483 Instead of closing it never at all. Previously there was only code that
1484 closed the client asynchronously if sending the response happened
1485 asynchrously at a later time.
1486 Thanks to Christian M for debugging this issue.
1487 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1488 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1490 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1492 * gst/rtsp-server/rtsp-stream.c:
1493 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1494 Otherwise no sink is found for multicast sreams and the less accurate
1495 fallback is used to determine the current sequence number and timestamp.
1497 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1499 * gst/rtsp-server/rtsp-auth.c:
1500 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1501 When using the basic authentication scheme, we wouldn't validate that
1502 the authorization field of the credentials is not NULL and pass it on
1503 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1504 dereference the NULL pointer and crash.
1505 A specially crafted (read: invalid) RTSP header can cause this to
1507 As a solution, check for the authorization to be not NULL before
1508 continuing processing it and if it is simply fail authentication.
1509 This fixes CVE-2020-6095 and TALOS-2020-1018.
1510 Discovered by Peter Wang of Cisco ASIG.
1512 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1514 * gst/rtsp-server/rtsp-client.c:
1515 rtsp-client: Use watch_context before unref
1516 Move the usage of priv->watch_context to beginning of function
1517 gst_rtsp_client_finalize. Instead of use it after
1518 g_main_context_unref (priv->watch_context).
1520 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1522 * gst/rtsp-server/rtsp-stream.c:
1523 rtsp-stream: fix deadlock on transport removal
1524 We cannot take the RTSPStream lock while holding a transport backlog
1525 lock, as remove_transport may be called externally, which will
1526 take first the RTSPStream lock then the transport backlog lock.
1528 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1530 * gst/rtsp-server/rtsp-server-internal.h:
1531 * gst/rtsp-server/rtsp-stream-transport.c:
1532 * gst/rtsp-server/rtsp-stream.c:
1533 rtsp-stream: clear backlog when removing transport
1534 This ensures we don't end up calling any of transports' callbacks
1535 with a potentially unreffed user_data (in practice, a client that
1536 may have been removed)
1538 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1540 * gst/rtsp-server/rtsp-stream.c:
1541 rtsp-stream: marshal calls to send_tcp_message to a single thread
1542 In order to address the race condition pointed out at
1543 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1544 we get rid of the send thread pool, and instead spawn and manage
1545 a single thread to pull samples from app sinks and add them to
1546 the transport's backlogs.
1547 Additionally, we now also always go through the backlogs in order
1548 to simplify the logic.
1550 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1552 * gst/rtsp-server/rtsp-server-internal.h:
1553 * gst/rtsp-server/rtsp-stream-transport.c:
1554 * gst/rtsp-server/rtsp-stream.c:
1555 rtsp-stream: properly protect TCP backlog access
1557 We cannot hold stream->lock while pushing data, but need
1558 to consistently check the state of the backlog both from
1559 the send_tcp_message function and the on_message_sent function,
1560 which may or may not be called from the same thread.
1561 This commit introduces internal API to allow for potentially
1562 recursive locking of transport streams, addressing a race
1563 condition where the RTSP stream could push items out of order
1564 when popping them from the backlog.
1566 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1568 * gst/rtsp-server/rtsp-media.c:
1569 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1570 It's taken ownership of by the media, and returned with `transfer none`
1571 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1572 first then any bindings will wrongly take ownership of the pipeline once
1573 it arrives in bindings code.
1575 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1577 * examples/test-onvif-client.c:
1578 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1580 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1582 * gst/rtsp-server/rtsp-media.c:
1583 rtsp-media: fix default latency
1585 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1587 * gst/rtsp-server/rtsp-client.c:
1588 rtsp-client: make closing more thread safe
1589 + Take the watch lock prior to using priv->watch
1590 + Flush both the watch and connection before closing / unreffing
1591 gst_rtsp_connection_close() is not threadsafe on its own, this is
1592 a workaround at the client level, where we control both the watch
1595 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1597 * gst/rtsp-server/rtsp-latency-bin.c:
1598 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1601 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1602 `your_type_get_instance_private()` function instead
1605 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1607 * gst/rtsp-server/rtsp-client.c:
1608 * tests/check/gst/rtspserver.c:
1609 rtsp-client: add property post-session-timeout
1610 This is a TCP connection timeout for client connections, in seconds.
1611 If a positive value is set for this property, the client connection
1612 will be kept alive for this amount of seconds after the last session
1613 timeout. For negative values of this property the connection timeout
1614 handling is delegated to the system (just as it was before).
1617 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1619 * gst/rtsp-server/rtsp-stream.c:
1620 rtsp-stream: check for NULL transports prior to ref'ing
1622 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1624 * gst/rtsp-server/rtsp-server-internal.h:
1625 * gst/rtsp-server/rtsp-stream-transport.c:
1626 * gst/rtsp-server/rtsp-stream.c:
1627 rtsp-stream: fix checking of TCP backpressure
1628 The internal index of our appsinks, while it can be used to
1629 determine whether a message is RTP or RTCP, is not necessarily
1630 the same as the interleaved channel. Let the stream-transport
1631 determine the channel to check backpressure for, the same way
1632 it determines the channel according to whether it is sending
1635 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1637 * gst/rtsp-server/rtsp-session.c:
1638 rtsp-session: Butcher the file to please gst-indent in the CI
1639 This should be reverted once the CI has an updated gst-indent.
1641 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1643 * gst/rtsp-server/rtsp-session.c:
1644 * gst/rtsp-server/rtsp-session.h:
1645 * gst/rtsp-sink/gstrtspclientsink.c:
1646 * gst/rtsp-sink/gstrtspclientsink.h:
1647 rtsp-session & client: Remove deprecated GTimeVal
1648 GTimeVal won't work past 2038
1650 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1652 * gst/rtsp-server/rtsp-auth.c:
1653 rtsp-auth: fix default token leak
1655 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1657 * gst/rtsp-sink/gstrtspclientsink.c:
1658 gstrtspclientsink: unref transports when closing bin
1661 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1663 * gst/rtsp-server/rtsp-media.c:
1664 rtsp-media: Force seek when flush flag is set
1665 The commit "rtsp-client: define all seek accuracy flags from
1666 setup_play_mode" changed the behaviour of when doing a seek.
1667 Before that commit, having the flush flag set would result in a seek
1669 Even if no seek was needed. One reason to force seek is to flush old buffers
1670 created in Describe requests.
1671 Thus adding force seek also for flush flag will result in play request
1674 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1676 * gst/rtsp-server/rtsp-client.c:
1677 rtsp-client: Revitalize dead code
1678 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1681 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1683 * gst/rtsp-server/rtsp-sdp.c:
1684 rtsp-sdp: Don't try to use non-initialized values
1685 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1686 returns TRUE. Also avoid the whole clock signalling block if we're not
1687 dealing with senders.
1692 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1694 * gst/rtsp-server/rtsp-stream-transport.c:
1695 * gst/rtsp-server/rtsp-stream.c:
1696 * tests/check/gst/stream.c:
1697 rtsp-stream: Removing invalid transports returns false
1698 When removing transports an assertion was that the transports passed in
1699 for removal are present in the list, however that can't be assumed.
1700 As an example if a transport was removed from a thread running
1701 send_tcp_message, the main thread can try to remove the same transport
1702 again if it gets a handle_pause_request. This will not effect the
1703 transport list but it will effect n_tcp_transports as it will be
1704 decrement and then have the wrong value.
1706 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1708 * tests/check/gst/client.c:
1709 client test: add scale and speed negative tests
1710 Negative tests for scale and speed should be done as well, verify that
1711 the response code is "400 Bad request" when a bad request is done.
1713 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1715 * gst/rtsp-server/rtsp-auth.c:
1716 * gst/rtsp-server/rtsp-client.c:
1717 * gst/rtsp-server/rtsp-media-factory.c:
1718 * gst/rtsp-server/rtsp-media.c:
1719 * gst/rtsp-server/rtsp-server.c:
1720 * gst/rtsp-server/rtsp-session-pool.c:
1721 * gst/rtsp-server/rtsp-stream.c:
1722 * gst/rtsp-sink/gstrtspclientsink.c:
1723 Don't pass default GLib marshallers for signals
1724 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1725 actually internally optimize the signal (if the marshaller is available
1726 in GLib itself) by also setting the valist marshaller. This makes the
1727 signal emission a bit more performant than the regular marshalling,
1728 which still needs to box into `GValue` and call libffi in case of a
1730 Note that for custom marshallers, one would use
1731 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1733 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1735 * gst/rtsp-server/rtsp-mount-points.c:
1736 GstRTSPMountPoints: Remove any existing factory before adding a new one
1737 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1738 previous mount point will be freed" which was true when it was
1739 implemented using a GHashTable. But in 2012 it got rewrote using a
1740 GSequence and since then it could have 2 factories for the same path.
1741 Which one gets used is random, depending on the sorting order of 2
1744 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1746 * gst/rtsp-server/rtsp-client.c:
1747 * gst/rtsp-server/rtsp-server-internal.h:
1748 * gst/rtsp-server/rtsp-stream-transport.c:
1749 * gst/rtsp-server/rtsp-stream-transport.h:
1750 * gst/rtsp-server/rtsp-stream.c:
1751 stream: refactor TCP backpressure handling
1752 The previous implementation stopped sending TCP messages to
1753 all clients when a single one stopped consuming them, which
1754 obviously created problems for shared media.
1755 Instead, we now manage a backlog in stream-transport, and slow
1756 clients are removed once this backlog exceeds a maximum duration,
1757 currently hardcoded.
1760 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1763 meson: build gir even when cross-compiling if introspection was enabled explicitly
1764 This can be made to work in certain circumstances when
1765 cross-compiling, so default to not building g-i stuff
1766 when cross-compiling, but allow it if introspection was
1767 enabled explicitly via -Dintrospection=enabled.
1768 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1770 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1772 * gst/rtsp-server/rtsp-session.c:
1773 rtsp-session: clean up comment extra-timeout
1775 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1777 * gst/rtsp-server/rtsp-client.c:
1778 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1779 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1780 from the RTSP context.
1783 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1785 * gst/rtsp-server/rtsp-client.c:
1786 * gst/rtsp-server/rtsp-media.c:
1787 * gst/rtsp-server/rtsp-media.h:
1788 rtsp-client: Lock shared media
1789 For shared media we got race conditions. Concurrently rtsp clients might
1790 suspend or unsuspend the shared media and thus change the state without
1791 the clients expecting that.
1792 By introducing a lock that can be taken by callers such as rtsp_client
1793 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1794 to handle the media sequentially thus allowing one client to finish its
1795 rtsp call before another client calls on the same media.
1796 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1799 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1801 * gst/rtsp-server/rtsp-session.c:
1802 rtsp-session: add property extra-timeout
1803 Extra time to add to the timeout, in seconds. This only
1804 affects the time until a session is considered timed out
1805 and is not signalled in the RTSP request responses.
1806 Only the value of the timeout property is signalled in the
1809 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1811 * gst/rtsp-server/rtsp-stream.c:
1812 rtsp-stream : fix race condition in send_tcp_message
1813 If one thread is inside the send_tcp_message function and are done
1814 sending rtp or rtcp messages so the n_outstanding variable is zero
1815 however have not exit the loop sending the messages. While sending its
1816 messages, transports have been added or removed to the transport list,
1817 so the cache should be updated. If now an additional thread comes to
1818 the function send_tcp_message and trying to send rtp messages it will
1819 first destroy the rtp cache that is still being iterated trough by the
1823 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1832 * examples/.gitignore:
1833 * examples/Makefile.am:
1835 * gst/rtsp-server/.gitignore:
1836 * gst/rtsp-server/Makefile.am:
1837 * gst/rtsp-sink/Makefile.am:
1838 * pkgconfig/.gitignore:
1839 * pkgconfig/Makefile.am:
1841 * tests/Makefile.am:
1842 * tests/check/Makefile.am:
1843 Remove autotools build
1845 Maybe we can now use the meson pkgconfig module
1846 for .pc files? (Does it support uninstalled now?)
1848 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1850 * tests/check/gst/client.c:
1851 client: fix test mem leak in attach_rate_tweaking_probe
1853 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1855 * tests/check/gst/media.c:
1856 media: remove memleak in test test_media_seek
1858 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1860 * tests/check/gst/rtspserver.c:
1861 rtspserver: Remove memleak in test test_double_play
1863 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1865 * gst/rtsp-server/rtsp-media.c:
1866 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1868 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1870 * gst/rtsp-server/rtsp-media.c:
1871 * tests/check/gst/rtspserver.c:
1872 rtsp-media: Unblock all streams
1873 When unsuspending and going to PLAYING, unblock all streams instead of
1874 only those that are linked (the linked streams are the ones for which
1875 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1876 pushing buffers on unlinked streams.
1877 This change is because playback using single-threaded demuxers like
1878 matroska-demux could be blocked if SETUP was not called for all media.
1879 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1880 gstflvdemux, qtdemux, and matroska-demux) will handle
1881 GST_FLOW_NOT_LINKED automatically.
1884 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1886 * gst/rtsp-server/rtsp-media.c:
1887 * tests/check/gst/rtspserver.c:
1888 rtsp-media: Wait on async when needed.
1889 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1890 In the unit test the pause from adjust_play_mode will cause a preroll
1891 and after that async-done will be produced.
1892 Without this patch there are no one consuming this async-done and when
1893 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1894 wait for async-done. But then it wrongly find the async-done prodused by
1895 adjus_play_mode and continue executing without waiting for the preroll
1898 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1900 * gst/rtsp-server/rtsp-client.c:
1901 rtsp-client: RTP Info when completed_sender
1902 Change condition that should be fulfilled regarding RTPInfo.
1903 Replace !gst_rtsp_media_is_receive_only with
1904 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1905 for a sender pipeline that is complete. Only then a RTPInfo should
1907 gst_rtsp_media_is_receive_only gives different answears depending on
1909 If Describe is called wth URL+options for backchannel SDP will give only
1910 audio and only backchannel a=sendonly
1911 If Describe is called on URL+options that gives both audio and video
1912 direction from server to client, pipelines are created. Thus
1913 receive_only will return false, even though Setup only would setup
1915 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1916 streams are complete.
1918 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1920 * gst/rtsp-server/rtsp-client.c:
1921 * tests/check/gst/client.c:
1922 rtsp-client: RTP Info exists conditionally in PLAY
1923 If RTP Info is missing and it is not a receiver only, eg. audio
1924 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1925 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1926 Since 1.14 there is audio backchannel support. Thus RTP-info is
1927 conditional now. When audio backchannel only mode, there is no RTP-info.
1930 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1932 * examples/test-onvif-client.c:
1933 test-onvif-client: remove unused query
1935 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1937 * gst/rtsp-server/rtsp-client.c:
1938 rtsp-client: RTP Info must exist in PLAY response
1939 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1942 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1944 * examples/test-onvif-client.c:
1945 test-onvif-client: perform accurate seeks
1946 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1947 Also, modify how we compute the position: position queries in
1948 PAUSED mode fail to account for the newly-prerolled frame, leading
1949 to frame skips when performing seeks in that state. Instead,
1950 compute the current position from the last sample.
1952 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1954 * gst/rtsp-server/rtsp-client.c:
1955 * gst/rtsp-server/rtsp-media.c:
1956 * gst/rtsp-server/rtsp-media.h:
1957 * tests/check/gst/rtspserver.c:
1958 Use complete streams for scale and speed.
1959 Without this patch it's always stream0 that is used to get segment event
1960 that is used to set scale and speed. This even if client not doing SETUP
1961 for stream0. At least in suspend mode reset this not working since then
1962 it's just random if send_rtp_sink have got any segment event. There are
1963 no check if send_rtp_sink for stream0 got any data before media is
1964 prerolled after PLAY request.
1966 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1968 * examples/test-onvif-server.c:
1969 * examples/test-onvif-server.h:
1970 examples/onvif-server: fix werror build with clang
1971 ../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]
1972 self->incoming_segment->format, self->incoming_segment->flags,
1973 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1974 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1975 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1977 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1978 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1980 <scratch space>:77:1: note: expanded from here
1983 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1984 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1986 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1987 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1989 <scratch space>:9:1: note: expanded from here
1992 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1993 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1994 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1996 <scratch space>:12:1: note: expanded from here
2000 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
2003 meson: Don't generate doc cache when no plugins are enabled
2004 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
2006 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2008 * examples/test-onvif-client.c:
2009 test-onvif-client: stdin is not defined in MSVC
2011 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2013 * gst/rtsp-server/rtsp-media.c:
2014 rtsp-media: add missing Since tag
2016 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2018 * examples/test-onvif-client.c:
2019 test-onvif-client: STDIN_FILENO is not portable
2020 If not defined, define it to _fileno(stdin) on Windows, 0
2023 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2025 * examples/test-onvif-server.c:
2026 test-onvif-server: downgrade logging
2028 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2030 * examples/meson.build:
2031 * examples/test-onvif-client.c:
2032 * examples/test-onvif-server.c:
2033 examples: add ONVIF client / server example
2035 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2037 * gst/rtsp-server/rtsp-client.c:
2038 * gst/rtsp-server/rtsp-media.c:
2039 rtsp-client: define all seek accuracy flags from setup_play_mode
2040 We then pass those to adjust_play_mode, which needs to operate
2041 on the "final" seek flags, as previously the code in rtsp-media
2042 was assuming that accuracy seek flags (accurate / key_unit) should
2043 not be set if the flags passed to the seek method were already set.
2045 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
2047 * gst/rtsp-server/rtsp-media-factory-uri.c:
2048 * gst/rtsp-server/rtsp-media.c:
2049 rtsp-media: Try to get dynamic payloaders by name from their bin first
2050 First try "pay", then "pay_%s" (where %s == pad name). And only then
2051 fall back to the code that simply takes the first payloader that is
2053 The current code usually works (but is racy) because it will always take
2054 the payloader that was last added (due to g_list_prepend() when adding
2055 elements) in pad-added and that's usually the correct one. But if a new
2056 payloader is added between pad-added and us trying to get it, we would
2057 get the wrong payloader.
2059 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2061 * tests/check/gst/client.c:
2062 client test: expect any port in transport
2063 setup_multicast_client sets a 5000-5010 range for the client
2064 ports, it is incorrect to expect the transport to always use
2068 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2070 * tests/check/gst/onvif.c:
2071 onvif tests: use g_cond_wait() correctly
2072 g_cond_wait() has to be called in a loop until required conditions
2076 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
2078 * gst/rtsp-server/rtsp-stream.c:
2079 rtsp-stream: Not wait on receiver streams when pre-rolling
2080 Without this patch there are problem pre-rolling when using audio back
2082 Without this patch a probe will be created for all streams including
2083 the stream for audio backchannel. To pre-roll all this pads have to
2084 receive data. Since the stream for audio backchannel is a receiver this
2086 The solution is to never create any probes for streams that are for
2087 incomming data and instead set them as blocking already from beginning.
2089 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
2091 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2092 * gst/rtsp-server/rtsp-onvif-media.c:
2093 onvif-media: fix "void function returning a value" compiler warning
2095 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2097 * gst/rtsp-server/rtsp-media.c:
2098 rtsp-media: make sure streams are blocked when sending seek
2099 The recent ONVIF work exposed a race condition when dealing with
2100 multiple streams: one of the sinks may preroll before other streams
2101 have started flushing. This led to the pipeline posting async-done
2102 prematurely, when some streams were actually still in the middle
2103 of performing a flushing seek. The newly-added code looks up a
2104 sticky segment event on the first stream in order to respond to
2105 the PLAY request with accurate Scale and Speed headers. In the
2106 failure condition, the first stream was flushing, and thus had
2107 no sticky segment event, leading to the PLAY request failing,
2108 and in turn the test.
2110 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
2113 * gst/rtsp-server/rtsp-media-factory-uri.h:
2116 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2118 * gst/rtsp-server/rtsp-client.c:
2119 * gst/rtsp-server/rtsp-client.h:
2120 * gst/rtsp-server/rtsp-media.c:
2121 * gst/rtsp-server/rtsp-media.h:
2122 * gst/rtsp-server/rtsp-onvif-client.c:
2123 * gst/rtsp-server/rtsp-onvif-client.h:
2124 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2125 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2126 * gst/rtsp-server/rtsp-onvif-media.c:
2127 * gst/rtsp-server/rtsp-onvif-server.h:
2128 * gst/rtsp-server/rtsp-stream.c:
2129 * gst/rtsp-server/rtsp-stream.h:
2130 * tests/check/gst/media.c:
2131 * tests/check/gst/onvif.c:
2132 * tests/check/meson.build:
2133 onvif: Implement and test the Streaming Specification
2134 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2136 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2138 * gst/rtsp-server/rtsp-client.c:
2139 * gst/rtsp-server/rtsp-client.h:
2140 rtsp-client: add gst_rtsp_client_get_stream_transport()
2141 This will be used in the onvif tests in order to validate the
2142 data transmitted over TCP: for streaming to continue after a
2143 data message has been provided to client->send_func, the client
2144 is responsible for marking the message as sent on the relevant
2147 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2149 * gst/rtsp-server/rtsp-client.c:
2150 client: Scale implies TRICK_MODE
2152 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2154 * gst/rtsp-server/rtsp-client.c:
2155 client: compare booleans, not pointers to them
2157 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
2159 * gst/rtsp-server/rtsp-media.c:
2160 * gst/rtsp-server/rtsp-stream.c:
2161 * tests/check/gst/media.c:
2162 Reverse playback support
2163 GStreamer plays segment from stop to start when doing reverse playback.
2164 RTSP implies that media should be played from start of Range header to
2165 its stop. Hence we swap start and stop times before passing them to
2167 Also make gst_rtsp_stream_query_stop always return value that can be
2168 used as stop time of Range header.
2170 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
2172 * gst/rtsp-server/rtsp-client.c:
2173 * gst/rtsp-server/rtsp-media.c:
2174 * gst/rtsp-server/rtsp-media.h:
2175 * tests/check/gst/client.c:
2176 rtsp-client: add support for Scale and Speed header
2177 Add support for the RTSP Scale and Speed headers by setting the rate in
2178 the seek to (scale*speed). We then check the resulting segment for rate
2179 and applied rate, and use them as values for the Speed and Scale headers
2181 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2183 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
2185 * gst/rtsp-server/rtsp-client.c:
2186 * gst/rtsp-server/rtsp-client.h:
2187 rtsp-client: allow sub classes to adjust the seek
2188 Adds a new virtual function, adjust_play_mode(), that allows
2189 sub classes to adjust the seek done on the media. The sub class can
2190 modify the values of the the seek flags and the rate.
2191 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2193 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
2195 * gst/rtsp-server/rtsp-media.c:
2196 * gst/rtsp-server/rtsp-media.h:
2197 * gst/rtsp-server/rtsp-stream.c:
2198 * gst/rtsp-server/rtsp-stream.h:
2199 * tests/check/gst/media.c:
2200 rtsp-media: allow specifying rate when seeking
2201 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
2202 caller to specify the rate for the seek. Also added functions in
2203 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
2204 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2206 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
2210 meson: Bump minimal GLib version to 2.44
2211 This means we can use some newer features and get rid of some
2212 boilerplate code using the G_DECLARE_* macros.
2213 As discussed on IRC, 2.44 is old enough by now to start depending on it.
2215 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2217 * docs/libs/.gitignore:
2218 * docs/libs/Makefile.am:
2219 * docs/libs/gst-rtsp-server-docs.sgml:
2220 * docs/libs/gst-rtsp-server-sections.txt:
2221 * docs/libs/gst-rtsp-server.types:
2222 docs: remove obsolete gtk-doc related files
2224 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2226 * gst/rtsp-sink/gstrtspclientsink.c:
2227 doc: remove xml from comments
2229 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
2231 * docs/gst_plugins_cache.json:
2233 docs: Stop building the doc cache by default
2234 And update the cache
2235 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
2237 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
2239 * docs/gst_plugins_cache.json:
2240 docs: Update plugins documentation cache
2242 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
2245 * gst/rtsp-server/rtsp-context.c:
2246 * gst/rtsp-server/rtsp-session-pool.c:
2247 doc: Fix some docstrings
2249 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
2255 * docs/gst_plugins_cache.json:
2258 * docs/plugin-index.md:
2259 * docs/plugin-sitemap.txt:
2262 * docs/version.entities.in:
2263 * gst/rtsp-server/meson.build:
2264 * gst/rtsp-sink/meson.build:
2266 * meson_options.txt:
2267 docs: Port to hotdoc
2269 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2271 * gst/rtsp-server/rtsp-auth.c:
2272 * gst/rtsp-server/rtsp-client.h:
2273 rtsp-server: Fix various Since markers
2275 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
2277 * gst/rtsp-server/rtsp-media.c:
2278 * gst/rtsp-server/rtsp-sdp.c:
2279 * gst/rtsp-server/rtsp-session-media.c:
2280 * gst/rtsp-server/rtsp-stream.c:
2281 rtsp-server: Add various Since: 1.14 markers
2283 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
2285 * gst/rtsp-server/rtsp-media-factory.c:
2286 * gst/rtsp-server/rtsp-media.c:
2287 * gst/rtsp-server/rtsp-stream-transport.c:
2288 * gst/rtsp-server/rtsp-stream.c:
2289 rtsp-server: Add various missing Since: 1.16 markers
2291 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
2293 * gst/rtsp-sink/gstrtspclientsink.c:
2294 rtspclientsink: Set async-handling=false for the internal bins
2295 Without this we can easily run into a race condition with async state changes:
2296 - the pipeline is doing an async state change
2297 - we set the internal bins to PLAYING but that's ignored because an
2298 async state change is currently pending
2299 - the async state change finishes but does not change the state of the
2300 internal bins because of locked_state==TRUE
2301 - the internal bins stay in PAUSED forever
2303 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2305 * gst/rtsp-sink/gstrtspclientsink.c:
2306 rtspclientsink: Use write_messages() API to send buffer lists in one go
2307 And to write messages with multiple memories also via writev().
2309 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
2311 * gst/rtsp-server/rtsp-client.c:
2312 * gst/rtsp-server/rtsp-client.h:
2313 * gst/rtsp-server/rtsp-server-object.h:
2314 * gst/rtsp-server/rtsp-server.c:
2315 rtsp-client: Handle Content-Length limitation
2316 Add functionality to limit the Content-Length.
2317 API addition, Enhancement.
2318 Define an appropriate request size limit and reject requests
2319 exceeding the limit with response status 413 Request Entity Too Large
2322 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2329 === release 1.16.0 ===
2331 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2337 * gst-rtsp-server.doap:
2341 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
2343 * gst/rtsp-sink/gstrtspclientsink.c:
2344 rtspclientsink: Notify the stream transport about each written message
2345 Otherwise it will never try to send us the next one: it tries to keep
2346 exactly one message in-flight all the time.
2347 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
2348 in the client sink we always write data out synchronously.
2350 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
2352 * gst/rtsp-server/rtsp-stream.c:
2353 rtsp_server: Free thread pool before clean transport cache
2354 If not waiting for free thread pool before clean transport caches, there
2355 can be a crash if a thread is executing in transport list loop in
2356 function send_tcp_message.
2357 Also add a check if priv->send_pool in on_message_sent to avoid that a
2358 new thread is pushed during wait of free thread pool. This is possible
2359 since when waiting for free thread pool mutex have to be unlocked.
2361 === release 1.15.90 ===
2363 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
2369 * gst-rtsp-server.doap:
2373 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
2375 * gst/rtsp-server/rtsp-stream.c:
2376 rtsp-stream: Add support for GCM (RFC 7714)
2379 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2381 * gst/rtsp-server/rtsp-session-pool.c:
2382 session pool: fix missing klass-> in klass->create_session
2384 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2387 g-i: pass --quiet to g-ir-scanner
2388 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2389 that we get even if everything works just fine.
2390 We still get g-ir-scanner warnings and compiler warnings if
2391 we pass this option.
2393 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2396 g-i: silence 'nested extern' compiler warnings when building scanner binary
2397 We need a nested extern in our init section for the scanner binary
2398 so we can call gst_init to make sure GStreamer types are initialised
2399 (they are not all lazy init via get_type functions, but some are in
2400 exported variables). There doesn't seem to be any other mechanism to
2401 achieve this, so just remove that warning, it's not important at all.
2403 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2406 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2408 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2410 * gst/rtsp-server/rtsp-media.c:
2411 * tests/check/gst/media.c:
2412 rtsp-media: Handle set state when preparing.
2413 Handle the situation when a call to gst_rtsp_media_set_state is done
2414 when media status is preparing.
2415 Also add unit test for this scenario.
2416 The unit test simulate on a media level when two clients share a (live)
2418 Both clients have done SETUP and got responses. Now client 1 is doing
2419 play and client 2 is just closing the connection.
2420 Then without patch there are a problem when
2421 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2422 And client2 is doing closing connection we can end up in a call
2423 to gst_rtsp_media_set_state when
2424 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2425 shut down media is jumped over .
2426 With this patch and this scenario we wait until
2427 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2428 execute after that and now we will execute the logic for
2431 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2439 === release 1.15.2 ===
2441 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2447 * gst-rtsp-server.doap:
2451 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2453 * gst/rtsp-server/rtsp-media.c:
2454 * tests/check/gst/client.c:
2455 rtsp-media: Fix multicast use case with common media
2464 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2466 * gst/rtsp-server/rtsp-client.c:
2467 * gst/rtsp-server/rtsp-stream.c:
2468 * gst/rtsp-server/rtsp-stream.h:
2469 rtsp-server: remove recursive behavior
2470 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2472 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2474 * gst/rtsp-server/rtsp-client.c:
2475 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2476 And route all messages through the send_func if no send_messages_func
2478 We otherwise break backwards compatibility.
2480 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2482 * docs/libs/gst-rtsp-server-sections.txt:
2483 * gst/rtsp-server/rtsp-client.c:
2484 * gst/rtsp-server/rtsp-client.h:
2485 * gst/rtsp-server/rtsp-stream.c:
2486 rtsp-client: Add support for sending buffer lists directly
2487 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2489 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2491 * docs/libs/gst-rtsp-server-sections.txt:
2492 * gst/rtsp-server/rtsp-client.c:
2493 * gst/rtsp-server/rtsp-media.c:
2494 * gst/rtsp-server/rtsp-stream-transport.c:
2495 * gst/rtsp-server/rtsp-stream-transport.h:
2496 * gst/rtsp-server/rtsp-stream.c:
2497 * gst/rtsp-sink/gstrtspclientsink.c:
2498 rtsp-server: Add support for buffer lists
2499 This adds new functions for passing buffer lists through the different
2500 layers without breaking API/ABI, and enables the appsink to actually
2501 provide buffer lists.
2502 This should already reduce CPU usage and potentially context switches a
2503 bit by passing a whole buffer list from the appsink instead of
2504 individual buffers. As a next step it would be necessary to
2505 a) Add support for a vector of data for the GstRTSPMessage body
2506 b) Add support for sending multiple messages at once to the
2507 GstRTSPWatch and let it be handled internally
2508 c) Adding API to GOutputStream that works like writev()
2509 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2511 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2513 * gst/rtsp-server/rtsp-client.c:
2514 client: Fix crash in close handler
2515 The close handler could trigger a crash because it invalidated the
2516 watch_context while still leaving a source attached to it which would be
2517 cleaned up at a later point.
2519 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2521 * gst/rtsp-server/rtsp-stream.c:
2522 rtsp-stream: Use cached address when allocating sockets
2523 If an address/port was previously decided upon (ex: multicast in the
2524 SDP), then use that instead of re-creating another one
2525 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2527 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2529 * gst/rtsp-server/rtsp-media.c:
2530 rtsp-media: Fix race codition in finish_unprepare
2531 The previous fix for race condition around finish_unprepare where the
2532 function could be called twice assumed that the status wouldn't change
2533 during execution of the function. This assumption is incorrect as the
2534 state may change, for example if an error message arrives from the
2536 Instead a flag keeping track on whether the finish_unprepare function
2537 is currently executing is introduced and checked.
2538 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2540 === release 1.15.1 ===
2542 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2548 * gst-rtsp-server.doap:
2552 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2554 * gst/rtsp-server/rtsp-stream.c:
2555 Add source elements to the pipeline before activation
2556 In plug_src we changed the element state before adding it to
2557 the owner container. This prevented the pipeline from intercepting
2558 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2559 to assign a custom task pool.
2560 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2562 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2565 Automatic update of common submodule
2566 From ed78bee to 59cb678
2568 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2570 * examples/test-appsrc.c:
2571 examples: test-appsrc: fix coding style error
2573 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2575 * examples/test-appsrc.c:
2576 examples: test-appsrc: fix buffer leak
2578 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2580 * gst/rtsp-server/rtsp-media.c:
2581 rtsp-media: Update priv->blocked when linked streams are unblocked.
2582 Media is considered to be blocked when all streams that belong to
2583 that media are blocked.
2584 This patch solves the problem of inconsistent updates of
2585 priv->blocked that are not synchronized with the media state.
2587 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2589 * gst/rtsp-server/rtsp-media.c:
2590 rtsp-media: Don't block streams before seeking
2591 Before the seek operation is performed on media, it's required that
2592 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2593 At this stage, all transport parts (transport sinks) have been successfully
2594 added to the pipeline and there is no need for blocking the streams.
2596 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2598 * tests/check/gst/rtspserver.c:
2599 tests: rtspserver: Add shared media test case for TCP
2601 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2603 * gst/rtsp-server/rtsp-stream.c:
2604 rtsp-stream: Use seqnum-offset for rtpinfo
2605 The sequence number in the rtpinfo is supposed to be the first RTP
2606 sequence number. The "seqnum" property on a payloader is supposed to be
2607 the number from the last processed RTP packet. The sequence number for
2608 payloaders that inherit gstrtpbasepayload will not be correct in case of
2609 buffer lists. In order to fix the seqnum property on the payloaders
2610 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2611 "seqnum-offset" from the "stats" property contains the value of the
2612 very first RTP packet in a stream. The server will, however, try to look
2613 at the last simple in the sink element and only use properties on the
2614 payloader in case there no sink elements yet, and by looking at the last
2615 sample of the sink gives the server full control of which RTP packet it
2616 looks at. If the payloader does not have the "stats" property, "seqnum"
2617 is still used since "seqnum-offset" is only present in as part of
2618 "stats" and this is still an issue not solved with this patch.
2619 Needed for gst-plugins-base!17
2621 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2623 * gst/rtsp-server/rtsp-stream.c:
2624 rtsp-stream: Plug memory leak
2625 Attaching a GSource to a context will increase the refcount. The idle
2626 source will never be free'd since the initial reference is never
2629 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2632 Add Gitlab CI configuration
2633 This commit adds a .gitlab-ci.yml file, which uses a feature
2634 to fetch the config from a centralized repository. The intent is
2635 to have all the gstreamer modules use the same configuration.
2636 The configuration is currently hosted at the gst-ci repository
2637 under the gitlab/ci_template.yml path.
2638 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2640 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2643 * gst-rtsp-server.doap:
2644 Update git locations to gitlab
2646 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2648 * gst/rtsp-server/meson.build:
2649 meson: add new onvif types
2651 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2653 * gst/rtsp-server/meson.build:
2654 Add ONVIF subclass headers to the installed headers in meson.build too
2656 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2658 * gst/rtsp-server/rtsp-server-object.h:
2659 * gst/rtsp-server/rtsp-server.h:
2660 rtsp-server: Declare GstRTSPServer struct before anything else
2661 It's needed by all kinds of other headers, including the ones that are
2662 required for defining the GstRTSPServer struct itself and its API.
2664 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2666 * gst/rtsp-server/rtsp-onvif-client.h:
2667 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2668 * gst/rtsp-server/rtsp-onvif-media.h:
2669 * gst/rtsp-server/rtsp-onvif-server.h:
2670 Mark all ONVIF-specific subclasses as Since 1.14
2672 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2674 * gst/rtsp-server/Makefile.am:
2675 * gst/rtsp-server/meson.build:
2676 * gst/rtsp-server/rtsp-context.h:
2677 * gst/rtsp-server/rtsp-onvif-server.c:
2678 * gst/rtsp-server/rtsp-onvif-server.h:
2679 * gst/rtsp-server/rtsp-server-object.h:
2680 * gst/rtsp-server/rtsp-server-prelude.h:
2681 * gst/rtsp-server/rtsp-server.c:
2682 * gst/rtsp-server/rtsp-server.h:
2683 * gst/rtsp-server/rtsp-session.h:
2684 Include ONVIF types from single-include rtsp-server.h
2685 ... by actually making it a single-include header and moving everything
2686 related to the GstRTSPServer type to rtsp-server-object.h instead.
2687 Otherwise there are too many circular includes.
2688 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2690 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2692 * gst/rtsp-server/rtsp-client.c:
2693 * gst/rtsp-server/rtsp-latency-bin.c:
2694 * gst/rtsp-server/rtsp-stream.c:
2695 * gst/rtsp-server/rtsp-stream.h:
2696 rtsp-stream: use idle source in on_message_sent
2697 When the underlying layers are running on_message_sent, this sometimes
2698 causes the underlying layer to send more data, which will cause the
2699 underlying layer to run callback on_message_sent again. This can go on
2701 To break this chain, we introduce an idle source that takes care of
2702 sending data if there are more to send when running callback
2703 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2705 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2707 * gst/rtsp-server/rtsp-client.c:
2708 rtsp-client: Remove timeout GSource on cleanup
2709 Avoids ending up with races where a timeout would still be around
2710 *after* a client was gone. This could happen rather easily in
2711 RTSP-over-HTTP mode on a local connection, where each RTSP message
2712 would be sent as a different HTTP connection with the same tunnelid.
2713 If not properly removed, that timeout would then try to free again
2714 a client (and its contents).
2716 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2718 * gst/rtsp-server/Makefile.am:
2719 autotools: fix distcheck
2721 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2723 * gst/rtsp-server/Makefile.am:
2724 * gst/rtsp-server/meson.build:
2725 * gst/rtsp-server/rtsp-latency-bin.c:
2726 * gst/rtsp-server/rtsp-latency-bin.h:
2727 * gst/rtsp-server/rtsp-onvif-media.c:
2728 onvif: encapsulate onvif part into a bin
2729 ...and thus do not let onvif affect pipelines latency
2730 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2732 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2734 * tests/check/gst/client.c:
2735 tests: client: Avoid bind() failures in tests
2736 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2738 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2740 * gst/rtsp-server/rtsp-media-factory.c:
2741 * gst/rtsp-server/rtsp-media-factory.h:
2742 * gst/rtsp-server/rtsp-media.c:
2743 * gst/rtsp-server/rtsp-media.h:
2744 * gst/rtsp-server/rtsp-stream.c:
2745 * gst/rtsp-server/rtsp-stream.h:
2746 * tests/check/gst/client.c:
2747 * tests/check/gst/mediafactory.c:
2748 New property for socket binding to mcast addresses
2749 By default the multicast sockets are bound to INADDR_ANY,
2750 as it's not allowed to bind sockets to multicast addresses
2751 in Windows. This default behaviour can be changed by setting
2752 bind-mcast-address property on the media-factory object.
2753 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2755 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2758 * gst/rtsp-server/Makefile.am:
2759 * gst/rtsp-server/meson.build:
2760 * gst/rtsp-server/rtsp-address-pool.c:
2761 * gst/rtsp-server/rtsp-auth.c:
2762 * gst/rtsp-server/rtsp-client.c:
2763 * gst/rtsp-server/rtsp-context.c:
2764 * gst/rtsp-server/rtsp-media-factory-uri.c:
2765 * gst/rtsp-server/rtsp-media-factory.c:
2766 * gst/rtsp-server/rtsp-media.c:
2767 * gst/rtsp-server/rtsp-mount-points.c:
2768 * gst/rtsp-server/rtsp-params.c:
2769 * gst/rtsp-server/rtsp-permissions.c:
2770 * gst/rtsp-server/rtsp-sdp.c:
2771 * gst/rtsp-server/rtsp-server-prelude.h:
2772 * gst/rtsp-server/rtsp-server.c:
2773 * gst/rtsp-server/rtsp-session-media.c:
2774 * gst/rtsp-server/rtsp-session-pool.c:
2775 * gst/rtsp-server/rtsp-session.c:
2776 * gst/rtsp-server/rtsp-stream-transport.c:
2777 * gst/rtsp-server/rtsp-stream.c:
2778 * gst/rtsp-server/rtsp-thread-pool.c:
2779 * gst/rtsp-server/rtsp-token.c:
2781 libs: fix API export/import and 'inconsistent linkage' on MSVC
2782 Export rtsp-server library API in headers when we're building the
2783 library itself, otherwise import the API from the headers.
2784 This fixes linker warnings on Windows when building with MSVC.
2785 Fix up some missing config.h includes when building the lib which
2786 is needed to get the export api define from config.h
2787 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2789 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2791 * gst/rtsp-server/rtsp-media-factory.c:
2792 rtsp-media-factory: Add missing break statements
2793 This resulted in warnings/assertions whenever one accessed the
2794 max-mcast-ttl property.
2798 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2801 * meson_options.txt:
2802 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2804 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2807 * meson_options.txt:
2808 * tests/check/meson.build:
2809 meson: add option to disable build of rtspclientsink plugin
2811 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2813 * meson_options.txt:
2814 meson: re-arrange options
2816 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2819 * meson_options.txt:
2820 * tests/check/meson.build:
2821 * tests/meson.build:
2822 meson: Use feature option for tests option
2823 This was somehow missed the last time around.
2825 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2827 * gst/rtsp-server/meson.build:
2829 meson: Maintain macOS ABI through dylib versioning
2830 Requires Meson 0.48, but the feature will be ignored on older versions
2831 so it's safe to add it without bumping the requirement.
2833 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2835 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2837 * gst/rtsp-sink/meson.build:
2839 meson: add pkg-config file for the rtspclientsink plugin
2841 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2843 * gst/rtsp-server/rtsp-client.c:
2844 * tests/check/gst/client.c:
2845 rtsp-client: Avoid reuse of channel numbers for interleaved
2846 If a (strange) client would reuse interleaved channel numbers in
2847 multiple SETUP requests, we should not accept them. The channel
2848 numbers are used for looking up stream transports in the
2849 priv->transports hash table, and transports disappear from the table
2850 if channel numbers are reused.
2851 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2852 server to change the channel numbers suggested by the client.
2853 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2855 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2857 * tests/check/gst/client.c:
2858 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2859 Allow regex for matching transport header against expected pattern.
2860 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2862 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2864 * tests/check/meson.build:
2865 meson: There is no gstreamer-plugins-good-1.0.pc
2866 There is no installed version of that, only an uninstalled version.
2868 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2870 * gst/rtsp-server/rtsp-client.c:
2871 * tests/check/gst/stream.c:
2872 Fix indentation again
2874 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2876 * gst/rtsp-server/rtsp-client.c:
2877 * gst/rtsp-server/rtsp-stream.c:
2878 * gst/rtsp-server/rtsp-stream.h:
2879 * tests/check/gst/client.c:
2880 * tests/check/gst/stream.c:
2881 stream: Added a list of multicast client addresses
2882 When media is shared, the same media stream can be sent
2883 to multiple multicast groups. Currently, there is no API
2884 to retrieve multicast addresses from the stream.
2885 When calling gst_rtsp_stream_get_multicast_address() function,
2886 only the first multicast address is returned.
2887 With this patch, each multicast destination requested in SETUP
2888 will be stored in an internal list (call to
2889 gst_rtsp_stream_add_multicast_client_address()).
2890 The list of multicast groups requested by the clients can be
2891 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2892 There still exist some problems with the current implementation
2893 in the multicast case:
2894 1) The receiving part is currently only configured with
2895 regard to the first multicast client (see
2896 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2897 2) Secondly, of security reasons, some constraints should be
2898 put on the requested multicast destinations (see
2899 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2900 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2901 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2903 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2905 * gst/rtsp-server/rtsp-client.c:
2906 * gst/rtsp-server/rtsp-stream.c:
2907 * gst/rtsp-server/rtsp-stream.h:
2908 * tests/check/gst/client.c:
2909 stream: Choose the maximum ttl value provided by multicast clients
2910 The maximum ttl value provided so far by the multicast clients
2911 will be chosen and reported in the response to the current
2913 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2914 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2916 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2918 * gst/rtsp-server/rtsp-stream.c:
2919 * tests/check/gst/client.c:
2920 rtsp-stream: Don't require address pool in the transport specific case
2921 If "transport.client-settings" parameter is set to true, the client is
2922 allowed to specify destination, ports and ttl.
2923 There is no need for pre-configured address pool.
2924 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2925 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2927 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2929 * gst/rtsp-server/rtsp-client.c:
2930 * tests/check/gst/client.c:
2931 client: Don't reserve multicast address in the client setting case
2932 When two multicast clients request specific transport
2933 configurations, and "transport.client-settings" parameter is
2934 set to true, it's wrong to actually require that these two
2935 clients request the same multicast group.
2936 Removed test_client_multicast_invalid_transport_specific test
2937 cases as they wrongly require that the requested destination
2938 address is supposed to be present in the address pool, also in
2939 the case when "transport.client-settings" parameter is set to true.
2940 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2941 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2943 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2945 * gst/rtsp-server/rtsp-media-factory.c:
2946 * gst/rtsp-server/rtsp-media-factory.h:
2947 * gst/rtsp-server/rtsp-media.c:
2948 * gst/rtsp-server/rtsp-media.h:
2949 * gst/rtsp-server/rtsp-stream.c:
2950 * gst/rtsp-server/rtsp-stream.h:
2951 * tests/check/gst/mediafactory.c:
2952 Add new API for setting/getting maximum multicast ttl value
2953 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2954 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2956 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2958 * gst/rtsp-server/rtsp-stream.c:
2959 rtsp-stream: avoid duplicating the first multicast client
2960 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2961 clients were dynamically added and removed to the multicast
2962 udp sinks, as such we should no longer add a first client in
2963 set_multicast_socket_for_udpsink
2964 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2966 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2968 * gst/rtsp-server/rtsp-stream.c:
2969 Revert "rtsp-stream: avoid duplicating the first multicast client"
2970 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2971 Commits where accidentially squashed together
2973 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2975 * gst/rtsp-server/rtsp-client.c:
2976 * gst/rtsp-server/rtsp-media-factory.c:
2977 * gst/rtsp-server/rtsp-media-factory.h:
2978 * gst/rtsp-server/rtsp-media.c:
2979 * gst/rtsp-server/rtsp-media.h:
2980 * gst/rtsp-server/rtsp-stream.c:
2981 * gst/rtsp-server/rtsp-stream.h:
2982 * tests/check/gst/client.c:
2983 * tests/check/gst/mediafactory.c:
2984 Revert "Add new API for setting/getting maximum multicast ttl value"
2985 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2986 Commits where accidentially squashed together
2988 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2990 * gst/rtsp-server/rtsp-stream.c:
2991 * tests/check/gst/client.c:
2992 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2993 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2994 Commits where accidentially squashed together
2996 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2998 * gst/rtsp-server/rtsp-client.c:
2999 * gst/rtsp-server/rtsp-stream.c:
3000 * gst/rtsp-server/rtsp-stream.h:
3001 * tests/check/gst/client.c:
3002 * tests/check/gst/stream.c:
3003 Revert "stream: Choose the maximum ttl value provided by multicast clients"
3004 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
3005 Commits where accidentially squashed together
3007 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
3009 * examples/test-auth-digest.c:
3010 examples: Fix indentation
3012 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
3014 * gst/rtsp-server/rtsp-client.c:
3015 * gst/rtsp-server/rtsp-stream.c:
3016 * gst/rtsp-server/rtsp-stream.h:
3017 * tests/check/gst/client.c:
3018 * tests/check/gst/stream.c:
3019 stream: Choose the maximum ttl value provided by multicast clients
3020 The maximum ttl value provided so far by the multicast clients
3021 will be chosen and reported in the response to the current
3023 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3025 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3027 * gst/rtsp-server/rtsp-stream.c:
3028 * tests/check/gst/client.c:
3029 rtsp-stream: Don't require address pool in the transport specific case
3030 If "transport.client-settings" parameter is set to true, the client is
3031 allowed to specify destination, ports and ttl.
3032 There is no need for pre-configured address pool.
3033 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3035 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
3037 * gst/rtsp-server/rtsp-client.c:
3038 * gst/rtsp-server/rtsp-media-factory.c:
3039 * gst/rtsp-server/rtsp-media-factory.h:
3040 * gst/rtsp-server/rtsp-media.c:
3041 * gst/rtsp-server/rtsp-media.h:
3042 * gst/rtsp-server/rtsp-stream.c:
3043 * gst/rtsp-server/rtsp-stream.h:
3044 * tests/check/gst/client.c:
3045 * tests/check/gst/mediafactory.c:
3046 Add new API for setting/getting maximum multicast ttl value
3047 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3049 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3051 * gst/rtsp-server/rtsp-stream.c:
3052 rtsp-stream: avoid duplicating the first multicast client
3053 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
3054 clients were dynamically added and removed to the multicast
3055 udp sinks, as such we should no longer add a first client in
3056 set_multicast_socket_for_udpsink
3057 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3059 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
3061 * gst/rtsp-server/Makefile.am:
3062 rtsp-server: Add gstreamer-base gir dir in autotools
3064 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3066 * gst/rtsp-server/rtsp-client.c:
3067 * gst/rtsp-server/rtsp-stream.c:
3068 rtsp-client: always allocate both IPV4 and IPV6 sockets
3069 multiudpsink does not support setting the socket* properties
3070 after it has started, which meant that rtsp-server could no
3071 longer serve on both IPV4 and IPV6 sockets since the patches
3072 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
3074 When first connecting an IPV6 client then an IPV4 client,
3075 multiudpsink fell back to using the IPV6 socket.
3076 When first connecting an IPV4 client, then an IPV6 client,
3077 multiudpsink errored out, released the IPV4 socket, then
3078 crashed when trying to send a message on NULL nevertheless,
3079 that is however a separate issue.
3080 This could probably be fixed by handling the setting of
3081 sockets in multiudpsink after it has started, that will
3082 however be a much more significant effort.
3083 For now, this commit simply partially reverts the behaviour
3084 of rtsp-stream: it will continue to only create the udpsinks
3085 when needed, as was the case since the patches were merged,
3086 it will however when creating them, always allocate both
3087 sockets and set them on the sink before it starts, as was
3088 the case prior to the patches.
3089 Transport configuration will only error out if the allocation
3090 of UDP sockets fails for the actual client's family, this
3091 also downgrades the GST_ERRORs in alloc_ports_one_family
3092 to GST_WARNINGs, as failing to allocate is no longer
3094 https://bugzilla.gnome.org/show_bug.cgi?id=796875
3096 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3099 * meson_options.txt:
3100 meson: Convert common options to feature options
3101 These are necessary for gst-build to set options correctly. The
3102 remaining automagic option is cgroup support in examples.
3103 https://bugzilla.gnome.org/show_bug.cgi?id=795107
3105 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
3107 * gst/rtsp-server/rtsp-stream.c:
3108 rtsp-stream: Slightly simplify locking
3110 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
3112 * gst/rtsp-server/rtsp-client.c:
3113 * gst/rtsp-server/rtsp-stream-transport.c:
3114 * gst/rtsp-server/rtsp-stream-transport.h:
3115 * gst/rtsp-server/rtsp-stream.c:
3116 Limit queued TCP data messages to one per stream
3117 Before, the watch backlog size in GstRTSPClient was changed
3118 dynamically between unlimited and a fixed size, trying to avoid both
3119 unlimited memory usage and deadlocks while waiting for place in the
3120 queue. (Some of the deadlocks were described in a long comment in
3122 In the previous commit, we changed to a fixed backlog size of 100.
3123 This is possible, because we now handle RTP/RTCP data messages differently
3124 from RTSP request/response messages.
3125 The data messages are messages tunneled over TCP. We allow at most one
3126 queued data message per stream in GstRTSPClient at a time, and
3127 successfully sent data messages are acked by sending a "message-sent"
3128 callback from the GstStreamTransport. Until that ack comes, the
3129 GstRTSPStream does not call pull_sample() on its appsink, and
3130 therefore the streaming thread in the pipeline will not be blocked
3131 inside GstRTSPClient, waiting for a place in the queue.
3132 pull_sample() is called when we have both an ack and a "new-sample"
3133 signal from the appsink. Then, we know there is a buffer to write.
3134 RTSP request/response messages are not acked in the same way as data
3135 messages. The rest of the 100 places in the queue are used for
3136 them. If the queue becomes full of request/response messages, we
3137 return an error and close the connection to the client.
3138 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
3140 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
3142 * gst/rtsp-server/rtsp-client.c:
3143 rtsp-client: Use fixed backlog size
3144 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
3145 Preparation for the next commit, which changes to a different way of
3146 avoiding both deadlocks and unlimited memory usage with the watch
3149 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3151 * gst/rtsp-server/rtsp-media.c:
3152 rtsp-media: unref clock (if set) when finalizing
3153 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3155 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3157 * docs/libs/gst-rtsp-server-sections.txt:
3158 rtsp-media: add gst_rtsp_media_*_set_clock to docs
3159 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3161 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
3163 * gst/rtsp-server/rtsp-media-factory.c:
3164 media-factory: unref old clock when setting new clock
3165 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3167 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
3169 * gst/rtsp-server/rtsp-media-factory.c:
3170 media-factory: unref clock in finalize
3171 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3173 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
3175 * gst/rtsp-server/rtsp-onvif-media.c:
3176 rtsp-onvif-media: fix g-ir-scanner warnings
3178 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3181 .gitignore: add another example binary
3183 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
3185 * examples/meson.build:
3186 meson: add new test-appsrc2 example to meson build
3188 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
3190 * examples/Makefile.am:
3191 examples: fix build of new test-appsrc2 example
3192 Need to link against libgstapp-1.0.
3194 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
3196 * examples/.gitignore:
3197 * examples/Makefile.am:
3198 * examples/test-appsrc2.c:
3199 examples: Add test-appsrc2
3200 Add an example of feeding both audio and video into an RTSP
3201 pipeline via appsrc.
3203 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
3205 * gst/rtsp-server/rtsp-client.c:
3206 client: Strip transport parts as whitespaces could be around commas
3207 https://bugzilla.gnome.org/show_bug.cgi?id=758428
3209 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
3211 * gst/rtsp-server/rtsp-stream.c:
3212 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
3213 Fix race when setting up source elements.
3214 Since we set the source element(s) to PLAYING state before hooking
3215 them up to the downstream funnel, it's possible for the source element
3216 to receive packets before we actually get to linking it to the funnel,
3217 in which case buffers would be pushed out on an unlinked pad, causing
3218 it to error out and stop receiving more data.
3219 We fix this by blocking the source's srcpad until we have linked it.
3220 https://bugzilla.gnome.org/show_bug.cgi?id=796160
3222 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
3224 * gst/rtsp-server/rtsp-stream.c:
3225 rtsp-stream: Fix mismatch between allowed and configured protocols
3226 https://bugzilla.gnome.org/show_bug.cgi?id=796679
3228 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
3230 * gst/rtsp-server/rtsp-stream.c:
3231 rtsp-stream: Emit a signal when the SRTP decoder is created
3232 https://bugzilla.gnome.org/show_bug.cgi?id=778080
3234 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
3236 * gst/rtsp-server/rtsp-stream.c:
3237 rtsp-stream: Don't require presence of sinks in _get_*_socket()
3238 Transport specific sink elements are added to the pipeline
3239 in PLAY request and sockets are already created in SETUP so
3240 it's actually wrong to require the presence of sinks in
3241 _get_*_socket() functions.
3242 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3244 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
3246 * gst/rtsp-server/rtsp-stream.c:
3247 rtsp-stream: Update transport for multicast clients as well
3248 If a multicast client requests different transport settings
3249 than the existing one make sure that this new transport
3250 configuruation is propagated to the multicast udp sink.
3251 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3253 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
3255 * gst/rtsp-server/rtsp-stream.c:
3256 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
3257 And not on unicast udp sinks
3258 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3260 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
3262 * gst/rtsp-server/rtsp-address-pool.c:
3263 * gst/rtsp-server/rtsp-auth.c:
3264 * gst/rtsp-server/rtsp-client.c:
3265 * gst/rtsp-server/rtsp-media-factory-uri.c:
3266 * gst/rtsp-server/rtsp-media-factory.c:
3267 * gst/rtsp-server/rtsp-media.c:
3268 * gst/rtsp-server/rtsp-mount-points.c:
3269 * gst/rtsp-server/rtsp-server.c:
3270 * gst/rtsp-server/rtsp-session-media.c:
3271 * gst/rtsp-server/rtsp-session-pool.c:
3272 * gst/rtsp-server/rtsp-session.c:
3273 * gst/rtsp-server/rtsp-stream-transport.c:
3274 * gst/rtsp-server/rtsp-stream.c:
3275 * gst/rtsp-server/rtsp-thread-pool.c:
3276 Update for g_type_class_add_private() deprecation in recent GLib
3278 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
3280 * gst/rtsp-server/rtsp-auth.c:
3281 * gst/rtsp-server/rtsp-media.c:
3282 * gst/rtsp-server/rtsp-sdp.c:
3283 * gst/rtsp-server/rtsp-stream.c:
3286 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
3288 * examples/Makefile.am:
3289 * examples/test-video-disconnect.c:
3290 examples: Add test-video-disconnect example
3291 Simple example which cuts off all clients 10 seconds
3292 after the first one connects.
3294 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3296 * docs/libs/gst-rtsp-server-sections.txt:
3297 * examples/test-auth-digest.c:
3298 * gst/rtsp-server/rtsp-auth.c:
3299 * gst/rtsp-server/rtsp-auth.h:
3300 rtsp-auth: Add support for parsing .htdigest files
3301 Passwords are usually not stored in clear text, but instead
3302 stored already hashed in a .htdigest file.
3303 Add support for parsing such files, add API to allow setting
3304 a custom realm in RTSPAuth, and update the digest example.
3305 https://bugzilla.gnome.org/show_bug.cgi?id=796637
3307 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
3309 * gst/rtsp-sink/gstrtspclientsink.c:
3310 * gst/rtsp-sink/gstrtspclientsink.h:
3311 rtspclientsink: fix waiting for multiple streams
3312 We were previously only ever waiting for a single stream to notify it's
3313 blocked status through GstRTSPStreamBlocking. Actually count streams to
3315 Fixes rtspclientsink sending SDP's without out some of the input
3317 https://bugzilla.gnome.org/show_bug.cgi?id=796624
3319 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3321 * docs/libs/gst-rtsp-server-sections.txt:
3322 docs: add missing auth methods
3324 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3326 * gst/rtsp-server/rtsp-stream.c:
3327 rtsp-stream: only create funnel if it didn't exist already.
3328 This precented using multiple protocols for the same stream.
3329 https://bugzilla.gnome.org/show_bug.cgi?id=796634
3331 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3333 * examples/meson.build:
3334 meson: build auth-digest example
3336 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
3338 * gst/rtsp-server/rtsp-client.c:
3339 * gst/rtsp-server/rtsp-media.c:
3340 * gst/rtsp-server/rtsp-sdp.c:
3341 * gst/rtsp-server/rtsp-session-media.c:
3342 * gst/rtsp-server/rtsp-stream-transport.c:
3343 Get payloader stats only for the sending streams
3344 Get/set payloader properties only for streams that actually
3345 contain a payloader element.
3346 https://bugzilla.gnome.org/show_bug.cgi?id=796523
3348 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
3350 * gst/rtsp-server/Makefile.am:
3351 Makefile: Don't hardcode libtool for g-i build
3352 Similar to the other commits in core/base/bad
3354 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
3356 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3357 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
3358 https://bugzilla.gnome.org/show_bug.cgi?id=796229
3360 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
3362 * gst/rtsp-sink/gstrtspclientsink.c:
3363 rtspclientsink: Don't deadlock in preroll on early close
3364 If the connection is closed very early, the flushing
3365 marker might not get set and rtspclientsink can get
3366 deadlocked waiting for preroll forever.
3367 https://bugzilla.gnome.org/show_bug.cgi?id=786961
3369 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3372 * meson_options.txt:
3373 meson: Update option names to omit disable_ and with- prefixes
3374 Also yield common options to the outer project (gst-build in our case)
3375 so that they don't have to be set manually.
3377 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3380 meson: use -Wl,-Bsymbolic-functions where supported
3381 Just like the autotools build.
3383 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3386 * tests/check/Makefile.am:
3387 configure: check for -good and -bad plugins only in uninstalled setup
3388 Avoids confusing configure messages looking or a -good .pc file
3390 Also use plugindir variables that common macros set while at it.
3391 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3393 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3395 * gst/rtsp-server/rtsp-client.c:
3396 rtsp-client: Fix session timeout
3397 When streaming data over TCP then is not the keep-alive
3398 functionality working.
3399 The reason is that the function do_send_data have changed
3400 to boolean but the code is still checking the received result
3401 from send_func with GST_RTSP_OK.
3402 The result is that a successful send_func will always lead to
3403 that do_send_data is returning false and the keep-alive will
3405 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3407 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3409 * docs/libs/gst-rtsp-server-sections.txt:
3410 * gst/rtsp-server/rtsp-media.c:
3411 * gst/rtsp-server/rtsp-sdp.c:
3412 * gst/rtsp-server/rtsp-stream.c:
3413 * gst/rtsp-server/rtsp-stream.h:
3414 * gst/rtsp-sink/gstrtspclientsink.c:
3415 * gst/rtsp-sink/gstrtspclientsink.h:
3416 Implement support for ULP Forward Error Correction
3417 In this initial commit, interface is only exposed for RECORD,
3418 further work will be needed in rtspsrc to support this for
3420 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3422 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3424 * gst/rtsp-server/rtsp-onvif-media.c:
3425 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3426 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3427 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3428 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3429 the opposite, just like the ONVIF standard.
3430 Let's follow those RFCs as we're doing RTSP here, and add a property at
3431 a later time if needed to switch to the SDP RFC behaviour.
3432 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3434 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3437 Automatic update of common submodule
3438 From 3fa2c9e to ed78bee
3440 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3442 * gst/rtsp-server/rtsp-client.c:
3443 * gst/rtsp-server/rtsp-media-factory.c:
3444 * gst/rtsp-server/rtsp-media.c:
3445 * gst/rtsp-server/rtsp-stream.c:
3446 * tests/check/gst/rtspclientsink.c:
3447 gst: Run everything through gst-indent again
3449 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3451 * gst/rtsp-server/rtsp-media.c:
3452 * tests/check/gst/media.c:
3453 rtsp-media: query the position on active streams if media is complete
3454 If the media is complete, i.e. one or more streams have been configured
3455 with sinks, then we want to query the position on those streams only.
3456 A query on an incomplete stream may return a position that originates from
3458 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3460 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3462 * gst/rtsp-sink/gstrtspclientsink.c:
3463 rtspclientsink: make sure not to use freed string
3464 Set transport string to NULL after freeing it, so that
3465 at worst we get a NULL pointer if constructing a new
3466 transport string fails (which shouldn't really fail here).
3467 Also check return value of that, just in case.
3470 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3472 * gst/rtsp-server/rtsp-client.c:
3473 rtsp-client: do not free string passed to take_header
3475 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3477 * gst/rtsp-server/rtsp-stream.c:
3478 rtsp-stream: do not take lock in request_aux_receiver
3479 Added it right before pushing the previous commit, it is
3480 incorrect and deadlocks because this function gets called
3481 from the join_bin thread, which already holds the lock,
3482 that's the reason why request_aux_sender didn't take the
3485 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3487 * docs/libs/gst-rtsp-server-sections.txt:
3488 * gst/rtsp-server/rtsp-media-factory.c:
3489 * gst/rtsp-server/rtsp-media-factory.h:
3490 * gst/rtsp-server/rtsp-media.c:
3491 * gst/rtsp-server/rtsp-media.h:
3492 * gst/rtsp-server/rtsp-stream.c:
3493 * gst/rtsp-server/rtsp-stream.h:
3494 rtsp-server: add API to enable retransmission requests
3495 "do-retransmission" was previously set when rtx-time != 0,
3496 which made no sense as do-retransmission is used to enable
3497 the sending of retransmission requests, where as rtx-time
3498 is used by the peer to enable storing of buffers in order
3499 to respond to retransmission requests.
3500 rtsp-media now also provides a callback for the
3501 request-aux-receiver signal.
3502 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3504 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3506 * gst/rtsp-sink/gstrtspclientsink.c:
3507 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3508 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3510 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3512 * gst/rtsp-sink/gstrtspclientsink.c:
3513 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3514 This in order to be able to decrypt the RTCP backchannel
3515 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3517 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3519 * gst/rtsp-server/rtsp-client.c:
3520 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3521 When sending back an encrypted RTCP back channel, it is useful
3522 for the client to know the encryption key.
3523 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3525 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3527 * gst/rtsp-server/rtsp-client.c:
3528 * gst/rtsp-server/rtsp-stream.c:
3529 * gst/rtsp-server/rtsp-stream.h:
3530 rtsp-stream: extract handle_keymgmt from rtsp-client
3531 rtspclientsink will also need to parse KeyMgmt headers
3532 sent by the server to decrypt the RTCP backchannel stream
3533 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3535 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3537 * gst/rtsp-sink/gstrtspclientsink.c:
3538 * tests/check/gst/rtspclientsink.c:
3539 rtspclientsink: Fix client ports for the RTCP backchannel
3540 This was broken since the work for delayed transport creation
3541 was merged: the creation of the transports string depends on
3542 calling stream_get_server_port, which only starts returning
3543 something meaningful after a call to stream_allocate_udp_sockets
3544 has been made, this function expects a transport that we parse
3545 from the transport string ...
3546 Significant refactoring is in order, but does not look entirely
3547 trivial, for now we put a band aid on and create a second transport
3548 string after the stream has been completed, to pass it in
3549 the request headers instead of the previous, incomplete one.
3550 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3552 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3554 * gst/rtsp-server/rtsp-client.c:
3555 rtsp-client:Error handling when equal http session cookie
3556 There are some clients that are sending same session cookie on random
3558 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3560 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3562 * gst/rtsp-server/rtsp-media-factory-uri.c:
3563 rtsp-media-factory-uri: Fix compilation with latest GLib
3564 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3565 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3566 data->factory = g_object_ref (factory);
3569 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3577 === release 1.14.0 ===
3579 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3585 * gst-rtsp-server.doap:
3589 === release 1.13.91 ===
3591 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3597 * gst-rtsp-server.doap:
3601 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3603 * gst/rtsp-server/Makefile.am:
3604 * gst/rtsp-server/meson.build:
3605 * gst/rtsp-server/rtsp-address-pool.h:
3606 * gst/rtsp-server/rtsp-auth.h:
3607 * gst/rtsp-server/rtsp-client.h:
3608 * gst/rtsp-server/rtsp-context.h:
3609 * gst/rtsp-server/rtsp-media-factory-uri.h:
3610 * gst/rtsp-server/rtsp-media-factory.h:
3611 * gst/rtsp-server/rtsp-media.h:
3612 * gst/rtsp-server/rtsp-mount-points.h:
3613 * gst/rtsp-server/rtsp-onvif-client.h:
3614 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3615 * gst/rtsp-server/rtsp-onvif-media.h:
3616 * gst/rtsp-server/rtsp-onvif-server.h:
3617 * gst/rtsp-server/rtsp-params.h:
3618 * gst/rtsp-server/rtsp-permissions.h:
3619 * gst/rtsp-server/rtsp-sdp.h:
3620 * gst/rtsp-server/rtsp-server-prelude.h:
3621 * gst/rtsp-server/rtsp-server.h:
3622 * gst/rtsp-server/rtsp-session-media.h:
3623 * gst/rtsp-server/rtsp-session-pool.h:
3624 * gst/rtsp-server/rtsp-session.h:
3625 * gst/rtsp-server/rtsp-stream-transport.h:
3626 * gst/rtsp-server/rtsp-stream.h:
3627 * gst/rtsp-server/rtsp-thread-pool.h:
3628 * gst/rtsp-server/rtsp-token.h:
3629 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3630 We need different export decorators for the different libs.
3631 For now no actual change though, just rename before the release,
3632 and add prelude headers to define the new decorator to GST_EXPORT.
3634 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3636 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3637 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3638 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3640 === release 1.13.90 ===
3642 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3648 * gst-rtsp-server.doap:
3652 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3654 * gst/rtsp-server/rtsp-media-factory.c:
3655 * gst/rtsp-server/rtsp-permissions.c:
3656 permissions: add Since tags and example for new API
3658 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3660 * docs/libs/gst-rtsp-server-sections.txt:
3661 * gst/rtsp-server/rtsp-media-factory.c:
3662 * gst/rtsp-server/rtsp-media-factory.h:
3663 * gst/rtsp-server/rtsp-permissions.c:
3664 * gst/rtsp-server/rtsp-permissions.h:
3665 * tests/check/gst/permissions.c:
3666 permissions: more bindings-friendly API
3667 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3669 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3672 meson: enable more warnings
3674 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3676 * gst/rtsp-server/rtsp-client.c:
3677 rtsp-client: Place netaddress meta on packets received via TCP
3678 This allows us to later map signals from rtpbin/rtpsource back to the
3679 corresponding stream transport, and allows to do keep-alive based on
3680 RTCP packets in case of TCP media transport.
3681 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3683 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3685 * gst/rtsp-sink/gstrtspclientsink.c:
3686 rtspclientsink: if OPEN failed, unqueue next command
3687 As READY_TO_PAUSED can no longer return async, the RECORD
3688 command will be queued before the OPEN command fails
3689 (for example in case the server could not be connected),
3690 and record then waits for ever.
3691 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3693 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3695 * gst/rtsp-sink/gstrtspclientsink.c:
3696 rtspclientsink: fix retrieval of custom payloader caps
3697 If a bin is passed as the custom payloader, the caps of
3698 its factory will be empty, the correct way to obtain the caps
3699 is to query its sinkpad.
3701 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3703 * gst/rtsp-sink/gstrtspclientsink.c:
3704 rtspclientsink: fix extra unref of custom payloader
3706 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3708 * gst/rtsp-sink/gstrtspclientsink.c:
3709 rspclientsink: fix recent code indentation
3711 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3713 * gst/rtsp-sink/gstrtspclientsink.c:
3714 rtspclientsink: add missing get_type prototype
3716 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3718 * gst/rtsp-sink/gstrtspclientsink.c:
3719 rtspclientsink: allow setting payloader as pad property
3720 This was a FIXME item, and can be quite useful, also
3721 allowing to specify payloader properties from the command
3722 line, which is always nice.
3723 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3725 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3727 * gst/rtsp-server/rtsp-media.c:
3728 rtsp-media: Replace g_print() log line
3729 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3731 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3733 * gst/rtsp-server/rtsp-media.c:
3734 * tests/check/gst/rtspclientsink.c:
3735 rtsp-media: fix RECORD getting stuck
3736 The test_record case was working because async=false had
3737 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3738 but that was incorrect, as it should not be needed.
3739 Removing async=false made the test fail as expected, this is
3740 fixed by not trying to preroll when preparing the media for
3741 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3742 and our peer will not start sending media until it has received
3743 a response to that request, and sent and received a response
3744 to RECORD as well, thus obviously preventing preroll.
3745 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3747 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3749 * gst/rtsp-server/rtsp-auth.c:
3750 rtsp-auth: fix set_tls_authentication_mode annotation
3752 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3754 * gst/rtsp-server/rtsp-onvif-media.c:
3755 rtp-server: remove redefined variable
3756 res is a boolean variable which is defined in the function scope and
3757 redefined, with no reason, in the loop scope. This patch removes the
3759 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3761 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3763 * gst/rtsp-server/rtsp-media.c:
3764 * gst/rtsp-server/rtsp-stream.c:
3765 * gst/rtsp-server/rtsp-stream.h:
3766 stream: Add functions for checking if stream is receiver or sender
3767 ...and replace all checks for RECORD in GstRTSPMedia which are really
3768 for "sender-only". This way the code becomes more generic and introducing
3769 support for onvif-backchannel later on will require no changes in
3772 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3774 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3775 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3776 onvif: Make requires_backchannel() public
3777 ...in order to let subclasses building the onvif part of the pipeline
3778 check whether backchannel shall be included or not.
3780 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3782 * gst/rtsp-server/rtsp-onvif-media.c:
3783 rtsp-server: Switch around sendonly/recvonly attributes
3784 They are wrong in the ONVIF streaming spec. The backchannel should be
3785 recvonly and the normal media should be sendonly: direction is always
3786 from the point of view of the SDP offerer (the server) according to
3789 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3791 * docs/libs/gst-rtsp-server-docs.sgml:
3792 * docs/libs/gst-rtsp-server-sections.txt:
3793 * examples/.gitignore:
3794 * examples/Makefile.am:
3795 * examples/test-onvif-backchannel.c:
3796 * gst/rtsp-server/Makefile.am:
3797 * gst/rtsp-server/rtsp-media.h:
3798 * gst/rtsp-server/rtsp-onvif-client.c:
3799 * gst/rtsp-server/rtsp-onvif-client.h:
3800 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3801 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3802 * gst/rtsp-server/rtsp-onvif-media.c:
3803 * gst/rtsp-server/rtsp-onvif-media.h:
3804 * gst/rtsp-server/rtsp-onvif-server.c:
3805 * gst/rtsp-server/rtsp-onvif-server.h:
3806 * gst/rtsp-server/rtsp-sdp.c:
3807 * gst/rtsp-server/rtsp-sdp.h:
3808 rtsp: Add support for ONVIF backchannel
3809 This adds a new RTSP server, client, media-factory and media subclass
3810 for handling the specifics of the backchannel. Ideally this later can be
3811 extended with other ONVIF specific features.
3813 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3815 * gst/rtsp-server/rtsp-media.c:
3816 rtsp-media: Add support for sending+receiving medias
3817 We need to add an appsrc/appsink in that case because otherwise the
3818 media bin will be a sink and a source for rtpbin, causing a pipeline
3820 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3822 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3828 === release 1.13.1 ===
3830 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3834 * gst-rtsp-server.doap:
3838 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3840 * gst/rtsp-server/rtsp-session-pool.c:
3841 session-pool: remove nullable return annotation
3842 create_watch can only return NULL from the API guards, no
3845 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3847 * gst/rtsp-server/rtsp-media-factory.c:
3848 * gst/rtsp-server/rtsp-media.c:
3849 set_clock functions: Add nullable annotations
3851 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3853 * gst/rtsp-server/rtsp-auth.c:
3854 * gst/rtsp-server/rtsp-client.c:
3855 * gst/rtsp-server/rtsp-media-factory.c:
3856 * gst/rtsp-server/rtsp-media.c:
3857 * gst/rtsp-server/rtsp-mount-points.c:
3858 * gst/rtsp-server/rtsp-server.c:
3859 * gst/rtsp-server/rtsp-session-media.c:
3860 * gst/rtsp-server/rtsp-session-pool.c:
3861 * gst/rtsp-server/rtsp-session.c:
3862 * gst/rtsp-server/rtsp-stream-transport.c:
3863 * gst/rtsp-server/rtsp-stream.c:
3864 * gst/rtsp-server/rtsp-thread-pool.c:
3865 All around: add annotations and API guards
3867 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3869 * tests/test-cleanup.c:
3870 test-cleanup: bind any port
3871 The meson test suite runs tests in parallel, trying to bind
3872 a single port made the test fail.
3874 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3877 meson: make version numbers ints and fix int/string comparison
3878 WARNING: Trying to compare values of different types (str, int).
3879 The result of this is undefined and will become a hard error
3880 in a future Meson release.
3882 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3884 * gst/rtsp-server/rtsp-context.c:
3885 gst_rtsp_context_get_current: add (skip) annotation
3886 The return value type is defined with G_DEFINE_POINTER_TYPE,
3887 and gi emits the following warning:
3888 Invalid non-constant return of bare structure or union; register as
3889 boxed type or (skip)
3891 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3893 * gst/rtsp-server/rtsp-client.c:
3894 rtsp-client: add type annotations
3895 gi doesn't seem to be able to figure out the type of the
3896 signal parameters when defined with G_DEFINE_POINTER_TYPE
3898 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3901 autotools: use -fno-strict-aliasing where supported
3902 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3904 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3907 meson: use -fno-strict-aliasing where supported
3908 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3910 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3912 * gst/rtsp-server/rtsp-mount-points.c:
3913 mount-points: bail out of loop again when matching mount points
3914 Previous patch led to us iterating the entire sequence. Bail out
3915 of the loop again if we have a match but are moving away from it.
3916 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3918 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3920 * tests/check/gst/mountpoints.c:
3921 tests: mountpoints: add more checks for mount point path matching
3922 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3924 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3926 * gst/rtsp-server/rtsp-mount-points.c:
3927 mount-points: fix matching of paths where there's also an entry with a common prefix
3928 e.g. with the following mount points
3932 _match() would not match /raw/video and /raw/snapshot correctly.
3933 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3935 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3937 * docs/libs/gst-rtsp-server-sections.txt:
3938 * gst/rtsp-server/rtsp-permissions.c:
3939 * gst/rtsp-server/rtsp-permissions.h:
3940 * tests/check/gst/permissions.c:
3941 permissions: add some new API to make this usable from bindings
3942 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3944 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3946 * gst/rtsp-server/rtsp-token.c:
3947 rtsp-token: annotate constructors for bindings
3948 This maps _new_empty() to _new(), which also makes RTSPToken()
3949 work properly now. Since this API wasn't usable from bindings
3950 before, this should hopefully be fine.
3951 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3953 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3955 * docs/libs/gst-rtsp-server-sections.txt:
3956 * gst/rtsp-server/rtsp-token.c:
3957 * gst/rtsp-server/rtsp-token.h:
3958 * tests/check/gst/token.c:
3959 rtsp-token: add some API to set fields from bindings
3960 The existing functions are all vararg-based and as such
3961 not usable from bindings.
3962 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3964 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3966 * tests/check/gst/rtspclientsink.c:
3967 * tests/check/gst/rtspserver.c:
3968 * tests/check/gst/sessionpool.c:
3969 * tests/check/gst/stream.c:
3970 tests: fix indentation
3973 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3975 * tests/check/gst/rtspserver.c:
3976 tests: rtspserver: fix another ref leak
3977 Even if this didn't show up in valgrind.
3979 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3981 * tests/check/gst/rtspclientsink.c:
3982 tests: rtspclientsink: fix leak
3984 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3986 * tests/check/gst/rtspserver.c:
3987 test: rtspserver: plug memory leak in test_no_session_timeout
3988 In test_no_session_timeout, unref the rtsp session object when the
3990 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3992 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3994 * gst/rtsp-sink/gstrtspclientsink.c:
3995 rtpsclientsink: Initialize and clear newly added mutex and cond
3996 While it *did* work, glib would automatically create new mutex and cond
3997 ... which never got freed
3999 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
4001 * gst/rtsp-server/rtsp-stream.c:
4002 rtsp-stream: Set multicast TTL on the multicast sockets
4003 And not if we do unicast UDP.
4004 https://bugzilla.gnome.org/show_bug.cgi?id=791743
4006 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
4008 * gst/rtsp-server/rtsp-stream.c:
4009 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
4010 In the multicast case (as in test-multicast, not test-multicast2), the
4011 address could be allocated/reserved (and thus set) already without
4012 allocating the actual socket. We need to allocate the socket here still
4013 instead of just claiming that it was already allocated.
4014 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
4016 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4018 * gst/rtsp-sink/gstrtspclientsink.c:
4019 * gst/rtsp-sink/gstrtspclientsink.h:
4020 rtspclientsink: Use the new rtsp-stream API
4021 https://bugzilla.gnome.org/show_bug.cgi?id=790412
4023 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4025 * gst/rtsp-sink/gstrtspclientsink.c:
4026 * gst/rtsp-sink/gstrtspclientsink.h:
4027 rtspclientsink: Wait until OPEN has been scheduled
4028 Make sure that the sink thread has started opening connection
4029 to the server before continuing.
4030 https://bugzilla.gnome.org/show_bug.cgi?id=790412
4032 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
4035 Automatic update of common submodule
4036 From e8c7a71 to 3fa2c9e
4038 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
4040 * gst/rtsp-server/rtsp-media.c:
4041 * gst/rtsp-server/rtsp-session-media.c:
4042 * gst/rtsp-server/rtsp-stream.c:
4043 rtsp-server: Minor doc fixes
4046 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4049 * tests/Makefile.am:
4050 tests: disable all tests when --disable-tests is used
4051 Move conditional subdir include into top level.
4052 Based on patch by: Joel Holdsworth
4053 https://bugzilla.gnome.org/show_bug.cgi?id=757703
4055 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
4058 * meson_options.txt:
4059 * tests/meson.build:
4060 meson: build more tests and add options to disable tests and examples
4062 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
4064 * gst/rtsp-server/rtsp-session.c:
4065 Fix build when -Werror=deprecated-declarations is on
4066 As gst_rtsp_session_next_timeout is deprecated.
4068 ../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]
4069 res = (gst_rtsp_session_next_timeout (session, now) == 0);
4071 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
4072 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
4073 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
4076 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
4079 Automatic update of common submodule
4080 From 3f4aa96 to e8c7a71
4082 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4084 * tests/check/gst/media.c:
4085 check/media: Add seekability test case: not all streams are active
4086 Media contains two streams but only one is complete and prepared
4088 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4090 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4092 * gst/rtsp-server/rtsp-stream.c:
4093 rtsp-stream: Do not reset 'blocking' if stream is already blocked
4094 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4096 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4098 * gst/rtsp-server/rtsp-media.c:
4099 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
4100 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4102 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
4105 meson: remove vs_module_defs_dir variable which is no longer needed
4107 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
4109 * gst/rtsp-server/rtsp-session.h:
4112 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
4115 * gst/rtsp-server/meson.build:
4117 * win32/common/libgstrtspserver.def:
4118 win32: remove .def file with exports
4119 They're no longer needed, symbol exporting is now explicit
4120 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
4122 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4125 autotools: stop controlling symbol visibility with -export-symbols-regex
4126 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
4127 This should result in consistent behaviour for the autotools and
4130 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4132 * gst/rtsp-server/rtsp-media.h:
4133 * gst/rtsp-server/rtsp-server.h:
4134 * gst/rtsp-server/rtsp-session.c:
4135 * gst/rtsp-server/rtsp-session.h:
4136 rtsp-server: add missing GST_EXPORT and export deprecated funcs
4138 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
4140 * tests/check/gst/media.c:
4141 check: Add seekability testing on medias
4142 Make sure that once GstRTSPMedia are prepared they returned
4143 the expected seekability results
4144 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4146 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
4148 * docs/libs/gst-rtsp-server-sections.txt:
4149 * gst/rtsp-server/rtsp-media.c:
4150 * gst/rtsp-server/rtsp-stream.c:
4151 * gst/rtsp-server/rtsp-stream.h:
4152 * win32/common/libgstrtspserver.def:
4153 rtsp-media: Enable seeking query before pipeline is complete
4154 SDP are now provided *before* the pipeline is fully complete. In order
4155 to know whether a media is seekable or not therefore requires asking
4156 the invididual streams.
4157 API: gst_rtsp_stream_seekable
4158 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4160 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
4162 * gst/rtsp-server/rtsp-media.c:
4163 rtsp-media: Fix handling in default_unsuspend()
4164 Handle the case when streams are not blocked and media
4165 is suspended from PAUSED.
4166 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
4167 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4169 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
4171 * tests/check/gst/media.c:
4172 check/media: Fix thread pool leak.
4173 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
4174 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4176 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
4178 * gst/rtsp-server/rtsp-media.c:
4179 rtsp-media: Removed fakesink elements
4180 There is not need of adding fakesink elements to the media
4181 pipeline in the dynamic-payloader case.
4182 The media pipeline itself is dynamically updated with
4183 the receiver and sender parts that are based on the client
4184 transport information known after SETUP has been received.
4185 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
4186 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4188 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
4190 * gst/rtsp-server/rtsp-media.c:
4191 rtsp-media: Corrected ASYNC_DONE handling
4192 Media is complete when all the transport based parts are
4193 added to the media pipeline. At this point ASYNC_DONE is
4194 posted by the media pipeline and media is ready to enter
4196 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
4197 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4199 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
4201 * tests/check/gst/media.c:
4202 check/media: Check that prepared media can provide a SDP
4203 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
4205 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
4207 * gst/rtsp-server/rtsp-client.c:
4208 rtsp-client: Don't leak addr
4211 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
4213 * gst/rtsp-server/rtsp-client.c:
4214 * gst/rtsp-server/rtsp-session-media.c:
4215 * gst/rtsp-server/rtsp-stream.c:
4218 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
4220 * gst/rtsp-server/rtsp-media.c:
4221 rtsp-media: Don't unblock with remaining dynamic payloaders
4222 If we still have some dynamic paylaoders which haven't posted
4223 no-more-pads yet, don't go to PREPARED if one of the streams
4225 The risk was that we would end up not exposing/using all specified
4227 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
4228 then it will take a bit more time to start. But only if those 3
4229 conditions are present.
4230 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4232 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
4234 * gst/rtsp-server/rtsp-media.c:
4237 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
4239 * gst/rtsp-server/rtsp-media.c:
4240 rtsp-media: Don't set float on a gint64 variable
4241 Just use 0. Fixes 'undefined' behaviour from clang
4243 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
4245 * gst/rtsp-server/rtsp-media.c:
4246 rtsp-media: Fix previous commit
4247 We only want to count dynamic payloaders
4249 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
4251 * gst/rtsp-server/rtsp-media.c:
4252 * tests/check/gst/media.c:
4253 rtsp-media: Handle multiple dynamic elements
4254 If we have more than one dynamic payloader in the pipeline, we need
4255 to wait until the *last* one emits 'no-more-pads' before switching
4257 Failure to do so would result in a race where some of the streams
4258 wouldn't properly be prepared
4259 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4261 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4263 * win32/common/libgstrtspserver.def:
4264 win32: Fix exported symbols list
4266 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
4268 * gst/rtsp-server/rtsp-stream.c:
4269 rtsp-stream: Only update the RTP udpsink if it actually exists
4270 For send-only streams it does not exist, but the RTCP udpsink might.
4272 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
4274 * win32/common/libgstrtspserver.def:
4275 win32: Update exports
4277 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
4279 * gst/rtsp-server/rtsp-media.c:
4280 * gst/rtsp-server/rtsp-stream.c:
4281 * gst/rtsp-server/rtsp-stream.h:
4282 rtsp-media: seek on media pipelines that are complete
4283 Make sure that a seek is performed on pipelines that
4284 contain at least one sink element.
4285 Change-Id: Icf398e10add3191d104b1289de612412da326819
4286 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4288 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
4290 * gst/rtsp-server/rtsp-client.c:
4291 * gst/rtsp-server/rtsp-media.c:
4292 * gst/rtsp-server/rtsp-media.h:
4293 * gst/rtsp-server/rtsp-stream.c:
4294 * gst/rtsp-server/rtsp-stream.h:
4295 * tests/check/gst/client.c:
4296 * tests/check/gst/media.c:
4297 * tests/check/gst/rtspserver.c:
4298 * tests/check/gst/stream.c:
4299 Dynamically reconfigure pipeline in PLAY based on transports
4300 The initial pipeline does not contain specific transport
4301 elements. The receiver and the sender parts are added
4303 If the media is shared, the streams are dynamically
4304 reconfigured after each PLAY.
4305 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4307 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
4309 * gst/rtsp-server/rtsp-stream.c:
4310 rtsp-stream: obtain stream position from pad
4311 If no sinks have been added yet, obtain the current and
4312 the stop position of the stream from the send_src pad.
4313 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
4314 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4316 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
4318 * gst/rtsp-server/rtsp-session-media.c:
4319 * gst/rtsp-server/rtsp-session-media.h:
4320 rtsp-session-media: add function to get a list of transports
4321 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
4322 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4324 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
4326 * gst/rtsp-server/rtsp-stream.c:
4327 * gst/rtsp-server/rtsp-stream.h:
4328 rtsp-stream: add functions to get rtp and rtcp multicast sockets
4329 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
4330 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4332 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
4334 * gst/rtsp-server/rtsp-stream.c:
4335 stream: set async=sync=false only for RTCP appsink
4336 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
4337 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4339 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
4341 * gst/rtsp-server/rtsp-media.c:
4342 rtsp-media: return minimum value in query position case
4343 The minimum position should be returned as we are interested
4344 in the whole interval.
4345 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
4346 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4348 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
4350 * gst/rtsp-server/rtsp-session.c:
4351 * tests/check/gst/rtspserver.c:
4352 rtsp-session: Handle the case when timeout=0
4353 According to the documentation, a timeout of value 0 means
4354 that the session never timeouts. This adds handling of that.
4355 If timeout=0 we just return with a -1 from
4356 gst_rtsp_session_next_timeout_usec ().
4357 https://bugzilla.gnome.org/show_bug.cgi?id=785058
4359 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4361 * gst/rtsp-sink/gstrtspclientsink.c:
4362 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
4363 https://bugzilla.gnome.org/show_bug.cgi?id=785024
4365 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
4367 * docs/libs/gst-rtsp-server-sections.txt:
4368 * gst/rtsp-server/rtsp-media-factory.c:
4369 docs: add media factory transport mode accessors
4370 and fix the documentation for the return value of the getter
4372 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
4374 * gst/rtsp-server/rtsp-client.c:
4375 rtsp-client: unref 'pipelined_requests' in finalize
4376 The hash table priv->pipelined_requests is not unref:ed in the
4377 finalize funktion. Make sure it is.
4378 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4380 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4382 * gst/rtsp-server/rtsp-media.c:
4383 rtsp-media: Initialize scalar variable
4386 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4388 * win32/common/libgstrtspserver.def:
4389 win32: Update export file
4391 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4393 * gst/rtsp-server/rtsp-client.c:
4394 * gst/rtsp-server/rtsp-media.c:
4395 * gst/rtsp-server/rtsp-media.h:
4396 Start support for RTSP 2.0
4397 This adds basic support for new 2.0 features, though the protocol is
4398 subposdely backward incompatible, most semantics are the sames.
4401 * version negotiation
4402 * pipelined requests support
4403 * Media-Properties support
4404 * Accept-Ranges support
4406 * gst_rtsp_media_seekable
4407 The RTSP methods that have been removed when using 2.0 now return
4409 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4411 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4413 * gst/rtsp-server/rtsp-stream.c:
4414 stream: Use stream duration as stream-stop if segment was not configured with a stop
4415 Allowing client to know stream duration when no seeking happened.
4416 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4418 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4420 * gst/rtsp-server/rtsp-media-factory.c:
4421 rtsp-media-factory: Don't cache any media if NULL was returned as key
4422 The docs already mentioned this, but we actually stored it in the hash
4423 table with key==NULL and leaked its reference forever.
4425 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4427 * gst/rtsp-sink/gstrtspclientsink.c:
4428 * gst/rtsp-sink/gstrtspclientsink.h:
4429 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4430 This is a simple port of:
4431 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4432 * c438545dc9e2f14f657bc0ef261fff726449867b
4433 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4434 in gst-plugins-good.
4436 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4438 * gst/rtsp-server/rtsp-sdp.c:
4439 sdp: fix Memory leak in error case
4440 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4442 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4444 * pkgconfig/meson.build:
4445 meson: don't install -uninstalled.pc file
4446 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4448 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4451 Automatic update of common submodule
4452 From 48a5d85 to 3f4aa96
4454 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4456 * gst/rtsp-server/rtsp-client.c:
4457 rtsp-client: Fix typo in debug message
4459 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4462 meson: hide symbols by default unless explicitly exported
4464 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4466 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4467 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4468 Fixes meson warning about undefined @srcdir@.
4470 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4472 * tests/meson.build:
4473 meson: skip tests on windows for now
4474 As we do in the other modules. As libgstcheck is currently not
4475 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4476 the subproject 'gstreamer' does not exist"" Meson error.
4478 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4480 * gst/rtsp-server/rtsp-stream.c:
4481 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4482 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4483 multiudpsink's last-sample always comes from the payloader. Which
4484 is wrong if auxiliary streams are multiplexed in the same stream.
4485 So check the buffer's ssrc against the caps'ssrc before to use its
4486 seqnum. If not the same ssrc just use the payloader as done prior
4487 the commit above or when there is no last-sample yet.
4488 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4490 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4493 meson: Allow using glib as a subproject
4495 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4498 meson: fix with-package-name option
4499 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4501 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4504 Distribute meson_options.txt
4506 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4509 And config.h.meson is no longer dist either
4511 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4515 meson: config.h.meson is no longer needed
4517 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4519 * tests/check/meson.build:
4520 * tests/meson.build:
4521 meson: Fix building tests and activate them again
4523 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4525 * tests/check/meson.build:
4526 meson: Do not use path separator in test names
4527 Avoiding warnings like:
4528 WARNING: Target "elements/audioamplify" has a path separator in its name.
4530 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4533 * meson_options.txt:
4534 meson: add options to set package name and origin
4535 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4537 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4539 * gst/rtsp-server/rtsp-address-pool.h:
4540 * gst/rtsp-server/rtsp-auth.h:
4541 * gst/rtsp-server/rtsp-client.h:
4542 * gst/rtsp-server/rtsp-context.h:
4543 * gst/rtsp-server/rtsp-media-factory-uri.h:
4544 * gst/rtsp-server/rtsp-media-factory.h:
4545 * gst/rtsp-server/rtsp-media.h:
4546 * gst/rtsp-server/rtsp-mount-points.h:
4547 * gst/rtsp-server/rtsp-params.h:
4548 * gst/rtsp-server/rtsp-permissions.h:
4549 * gst/rtsp-server/rtsp-sdp.h:
4550 * gst/rtsp-server/rtsp-server.h:
4551 * gst/rtsp-server/rtsp-session-media.h:
4552 * gst/rtsp-server/rtsp-session-pool.h:
4553 * gst/rtsp-server/rtsp-session.h:
4554 * gst/rtsp-server/rtsp-stream-transport.h:
4555 * gst/rtsp-server/rtsp-stream.h:
4556 * gst/rtsp-server/rtsp-thread-pool.h:
4557 * gst/rtsp-server/rtsp-token.h:
4558 Mark symbols explicitly for export with GST_EXPORT
4560 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4563 * gst/rtsp-sink/Makefile.am:
4564 Remove plugin specific static build option
4565 Static and dynamic plugins now have the same interface. The standard
4566 --enable-static/--enable-shared toggle are sufficient.
4568 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4574 === release 1.12.0 ===
4576 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4582 * gst-rtsp-server.doap:
4586 === release 1.11.91 ===
4588 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4594 * gst-rtsp-server.doap:
4598 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4601 Automatic update of common submodule
4602 From 60aeef6 to 48a5d85
4604 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4606 * gst/rtsp-server/rtsp-media-factory.c:
4607 * gst/rtsp-server/rtsp-media.c:
4608 * gst/rtsp-server/rtsp-session.c:
4609 * gst/rtsp-server/rtsp-stream.c:
4610 gi: Fix some annotations and docstrings
4612 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4614 * gst/rtsp-server/meson.build:
4616 * meson_options.txt:
4619 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4623 Automatic update of common submodule
4624 From 39ac2f5 to 60aeef6
4626 === release 1.11.90 ===
4628 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4634 * gst-rtsp-server.doap:
4638 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4640 * examples/test-launch.c:
4641 examples: make test-launch pipeline shared by default as well
4643 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4645 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4646 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4647 Just the build dir is not going to work for srcdir!=builddir.
4649 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4652 meson: Update version
4654 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4659 === release 1.11.2 ===
4661 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4667 * gst-rtsp-server.doap:
4670 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4673 meson: dist meson build files
4674 Ship meson build files in tarballs, so people who use tarballs
4675 in their builds can start playing with meson already.
4677 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4679 * examples/test-record.c:
4680 examples/test-record: Add extra line to initial printout
4681 Add an example line of how to deliver a stream to the
4684 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4686 * gst/rtsp-server/rtsp-client.c:
4687 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4688 If there is no Content-Length header, no body would be allocated and the
4689 '\0' would also not be appended to the body.
4691 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4693 * gst/rtsp-server/rtsp-client.c:
4694 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4695 While they logically have 0 bytes length, GstRTSPConnection is appending
4696 a '\0' to everything making the size be 1 instead.
4698 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4703 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4705 * gst/rtsp-server/rtsp-session.c:
4706 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4707 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4710 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4715 === release 1.11.1 ===
4717 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4723 * gst-rtsp-server.doap:
4724 * win32/common/libgstrtspserver.def:
4727 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4729 * gst/rtsp-server/rtsp-stream.c:
4730 rtsp-stream: corrected if-statement in _get_server_port()
4731 This bug was accidentally introduced while fixing a segfault
4732 in _get_server_port() function.
4733 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4735 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4737 * gst/rtsp-server/rtsp-stream.c:
4738 * tests/check/gst/stream.c:
4739 rtsp-stream: fixed segmenation fault in _get_server_port()
4740 Calling function gst_rtsp_stream_get_server_port() results in
4741 segmenation fault in the RTP/RTSP/TCP case.
4742 Port that the server will use to receive RTCP makes only
4743 sense in the UDP case, however the function should handle
4744 the TCP case in a nicer way.
4745 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4747 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4749 * gst/rtsp-server/rtsp-media-factory.c:
4750 dosc: Fix a little typo
4751 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4753 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4755 * pkgconfig/Makefile.am:
4756 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4757 * pkgconfig/meson.build:
4758 meson: generate pkg-config -uninstalled pc files
4759 Generating those files is useful for users building the GStreamer stack
4760 using meson and having to link it to another project which is still
4761 using the autotools.
4762 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4764 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4766 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4767 pkgconfig: fix -uninstalled pc file
4768 pcfiledir was never defined so the paths were wrong.
4769 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4771 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4773 * gst/rtsp-server/rtsp-stream.c:
4774 * tests/check/gst/rtspserver.c:
4775 rtsp-stream: Fixed TCP transport case
4776 Make sure that the appsink element is actually added to
4777 the bin before trying to link it with the elements in it.
4778 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4780 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4786 Remove generated .spec file
4787 Likely extremely bitrotten, and we should not ship this anyway.
4789 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4792 Automatic update of common submodule
4793 From f980fd9 to 39ac2f5
4795 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4797 * gst/rtsp-server/rtsp-media.c:
4798 media: Fix pt map caps
4799 Since decryption is handled within rtpbin, all outcoming stream
4800 caps will be application/x-rtp (i.e. regular rtp)
4801 Fixes RECORD with SRTP streams
4803 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4805 * gst/rtsp-server/rtsp-media-factory.c:
4806 media-factory: Create media objects with the proper transport mode
4807 The function called immediately afterwards (collect_streams()) will
4808 need it to work properly
4810 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4812 * gst/rtsp-server/rtsp-auth.c:
4813 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4815 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4817 * gst/rtsp-server/rtsp-media-factory.c:
4818 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4819 We're going to put a pipeline into a pipeline otherwise, which is not
4822 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4824 * gst/rtsp-server/rtsp-media.c:
4825 media: Fix race condition around finish_unprepare() if called multiple time
4826 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4828 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4830 * gst/rtsp-sink/gstrtspclientsink.c:
4831 rtspclientsink: Don't leave stale pointer after unref
4832 Fix a warning on shutdown - don't keep a pointer to an
4833 alread-unreffed object.
4835 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4838 common: use https protocol for common submodule
4839 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4841 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4843 * gst/rtsp-server/rtsp-stream.c:
4844 stream: block the output of rtpbin instead of the source pipeline
4845 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4846 detection of the srtp rollover counter to add to the SDP.
4847 Unfortunately, it was incomplete for live pipelines where the logic
4848 blocks the source bin before creating the SDP and thus would never have
4849 the necessary informaiton to create a correct SDP with srtp encryption.
4850 Move the pad blocks to rtpbin's output pads instead so that the
4851 necessary information can be created before we need the information for
4853 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4855 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4857 * gst/rtsp-server/rtsp-client.c:
4858 rtsp-client: add IDLE timeout, before session exists
4859 The RTSP server will not timeout an idle RTSP connection
4860 (note this is different from doing timeout on a RTSP
4862 At least for Apache this is a problem when running RTSP over
4863 HTTPS since it uses one of the threads (there is a rather
4864 limited number) that are available for handling requests.
4865 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4867 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4872 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4874 * gst/rtsp-server/rtsp-stream.c:
4875 rtsp-stream: Set close-socket FALSE on UDP src:es
4876 With this RTSP server can use the sockets independent on the udpsrc
4878 When the udp src is finalized it will unref socket and when g_socket
4879 is finalized the socket will be closed.
4880 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4882 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4884 * gst/rtsp-sink/gstrtspclientsink.c:
4885 rtspclientsink: Move to new helper function to parse authentication responses
4886 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4888 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4890 * examples/Makefile.am:
4891 * examples/test-auth-digest.c:
4892 * gst/rtsp-server/rtsp-auth.c:
4893 * gst/rtsp-server/rtsp-auth.h:
4894 * win32/common/libgstrtspserver.def:
4895 rtsp-auth: Add support for Digest authentication
4896 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4898 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4901 * gst/rtsp-server/meson.build:
4903 * tests/check/meson.build:
4905 * win32/common/libgstrtspserver.def:
4906 Enable building with MSVC
4907 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4909 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4912 meson: gstreamer gst_check_dep does not exist on windows
4914 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4916 * gst/rtsp-server/rtsp-client.c:
4917 client: update do_send_message to match type GstRTSPClientSendFunc
4918 This type mismatch fails building with MSVC
4919 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4921 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4923 * gst/rtsp-server/rtsp-sdp.c:
4924 rtsp-sdp: Fix indentation
4926 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4928 * gst/rtsp-server/rtsp-media.c:
4929 rtsp-media: Only signal "new-state" if the state has actually changed
4930 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4932 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4934 * gst/rtsp-server/rtsp-client.c:
4935 * gst/rtsp-server/rtsp-client.h:
4936 client: emit signal in the beginning of each rtsp request
4937 These signals let the application validate the requests, configure the
4938 media/stream in a certain way and also generate error status code in
4939 case of error or bad request.
4940 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4942 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4945 meson: update version
4947 === release 1.11.0 ===
4949 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4954 === release 1.10.0 ===
4956 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4962 * gst-rtsp-server.doap:
4965 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4967 * tests/check/gst/rtspserver.c:
4968 * tests/check/gst/stream.c:
4969 tests: try to avoid using the same ports in different tests
4970 Causes problems with client multicast tests otherwise if
4971 tests are run in parallel.
4972 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4974 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4976 * tests/check/gst/client.c:
4977 tests: client: use fail_unless_equals_foo() for better failure reporting
4979 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4981 * gst/rtsp-server/rtsp-client.c:
4982 rtsp-client: Session filter in unwatch session
4983 Call session filter with filter_session_media as paramer in
4984 client_unwatch_session if using drop_backlog = FALSE.
4985 In client_unwatch_session its allowed to grow the watchs backlog.
4986 If using drop_backlog = FALSE and the backlog is full it will cause
4987 a deadlock when setting session media state to NULL
4988 if the backlog is not allowed to grow.
4989 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4991 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4994 meson: add fallbacks for gst modules
4997 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4999 * gst/rtsp-server/rtsp-client.c:
5000 rtsp-client: Fix factory leaking in find_media() in error cases
5001 https://bugzilla.gnome.org/show_bug.cgi?id=771488
5003 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5005 * gst/rtsp-server/rtsp-stream.c:
5006 stream: Fix randomly missing streams from SDP with dynamic elements
5007 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
5008 "pad-added" signal. In that case priv->srcpad could already have its caps,
5009 and they'll be sent to priv->send_src[0] pad. That means that when it
5010 connects "notify::caps" signal, that pad could already have received its
5011 caps and the signal won't be emitted anymore.
5012 In that case priv->caps stay to NULL and when building the SDP that stream
5013 gets ignored. Leading to missing video or audio when playing in client side.
5014 https://bugzilla.gnome.org/show_bug.cgi?id=772478
5016 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
5019 meson: update version
5021 === release 1.9.90 ===
5023 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5029 * gst-rtsp-server.doap:
5032 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
5034 * gst/rtsp-server/rtsp-media-factory.c:
5035 * gst/rtsp-server/rtsp-media.c:
5036 * gst/rtsp-server/rtsp-stream.c:
5037 rtsp-server: Hint that set_multicast_iface expects the name of the interface
5038 To prevent any possibly confusion with IPs or anything else.
5039 https://bugzilla.gnome.org/show_bug.cgi?id=771530
5041 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
5043 * gst/rtsp-server/rtsp-media-factory.c:
5044 * gst/rtsp-server/rtsp-media.c:
5045 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
5046 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
5048 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
5051 configure: Depend on gstreamer 1.9.2.1
5053 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
5057 Automatic update of common submodule
5058 From b18d820 to f980fd9
5060 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
5064 Automatic update of common submodule
5065 From 6f2d209 to b18d820
5067 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
5069 * gst/rtsp-server/rtsp-stream.c:
5070 rtsp-stream: Remove unused _locked() variant of a function
5071 It was added during refactoring.
5073 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5075 * gst/rtsp-server/rtsp-stream.c:
5076 stream: cosmetic cleanup
5077 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5079 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5081 * gst/rtsp-server/rtsp-stream.c:
5082 stream: Compare IP addresses case insensitive in more places
5083 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5085 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5088 * gst/rtsp-server/rtsp-stream.c:
5089 stream: Fix leaked joined_bin
5090 There is no need to keep a strong ref on it, and _leave_bin() was
5091 setting it to NULL before calling g_clear_object() so it was leaked.
5092 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5094 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5096 * gst/rtsp-server/rtsp-stream.c:
5097 rtsp-stream: Compare IP address strings case insensitive
5098 Otherwise IPv6 addresses might fail this comparision.
5100 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
5102 * gst/rtsp-server/rtsp-stream.c:
5103 rtsp-stream: Bind multicast sockets to ANY as before
5104 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
5106 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
5108 * gst/rtsp-server/rtsp-session.c:
5109 rtsp-session: Fix segfault when doing keep-alive after removing the session
5110 If keep-alive happens after removing the session but before finalizing the
5111 stream transport, we would segfault.
5112 https://bugzilla.gnome.org/show_bug.cgi?id=750544
5114 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
5116 * gst/rtsp-server/rtsp-stream.c:
5117 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
5118 Adding them later will cause deadlocks due to
5119 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
5120 2) adding the multicast sink
5121 3) waiting for it to get data to preroll again
5122 3) never happens because the queues after the tee are full.
5124 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
5126 * gst/rtsp-server/rtsp-stream.c:
5127 rtsp-stream: Fix up various multicast related issues
5129 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
5131 * tests/check/gst/stream.c:
5132 tests: Fix compilation
5134 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5136 * gst/rtsp-server/rtsp-client.c:
5137 * gst/rtsp-server/rtsp-stream.c:
5138 * tests/check/gst/stream.c:
5139 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
5140 This is basically reverting changes introduced in commit f62a9a7,
5141 because it was introducing various regressions:
5142 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
5143 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
5144 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
5145 - If a mcast client connects, it creates a new socket in SETUP to try to respect
5146 the destination/port given by the client in the transport, and overrides the
5147 socket already set on the udpsink element. That means that if we already had a
5148 client connected, the source address on the udp packets it receives suddenly
5150 - If a 2nd mcast client connects, the destination/port in its transport is
5151 ignored but its transport wasn't updated.
5152 What this patch does:
5153 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
5154 - Always have a tee+queue when udp is enabled. This could be optimized
5155 again in a later patch, but is more complicated. If no unicast clients
5156 connects then those elements are useless, this could be also optimized
5158 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
5159 seperated from those for unicast clients. Since we already support only
5160 one mcast address, we also create only one set of elements.
5161 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5163 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5165 * gst/rtsp-server/rtsp-stream.c:
5166 stream: factor our plug_src function
5167 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5169 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5171 * gst/rtsp-server/rtsp-stream.c:
5172 stream: factor out plug_sink function
5173 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5175 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5177 * gst/rtsp-server/rtsp-stream.c:
5178 stream: small documentation clarification
5179 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5181 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5183 * gst/rtsp-server/rtsp-stream.c:
5184 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
5185 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5187 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5189 * gst/rtsp-server/rtsp-stream.c:
5190 stream: Keep a ref on joined bin
5191 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5193 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5195 * gst/rtsp-server/rtsp-stream.c:
5196 stream: code cleanup
5197 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5199 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5201 * gst/rtsp-server/rtsp-stream.c:
5202 stream: small fix in error code path
5203 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5205 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5207 * gst/rtsp-server/rtsp-stream.c:
5208 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
5209 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
5210 but keeps unit tests.
5211 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5213 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
5218 === release 1.9.2 ===
5220 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
5226 * gst-rtsp-server.doap:
5229 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
5232 * examples/meson.build:
5234 * gst/rtsp-server/meson.build:
5235 * gst/rtsp-sink/meson.build:
5237 * pkgconfig/meson.build:
5238 * tests/check/meson.build:
5239 * tests/meson.build:
5240 Add support for Meson as alternative/parallel build system
5241 https://github.com/mesonbuild/meson
5243 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
5246 * tests/check/Makefile.am:
5247 build: silence error about pthread for 'make check' in osx
5248 Fixes "clang: error: argument unused during compilation: '-pthread'"
5250 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
5252 * gst/rtsp-server/rtsp-client.c:
5253 rtsp-client: Fix leaking of media in error cases
5254 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
5255 and myself to make the media refcounting a bit easier to follow.
5256 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5258 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
5260 * gst/rtsp-server/rtsp-client.c:
5261 rtsp-client: Fix leaking of session in error cases
5262 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5264 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
5267 Automatic update of common submodule
5268 From f363b32 to f49c55e
5270 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
5275 === release 1.9.1 ===
5277 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5283 * gst-rtsp-server.doap:
5286 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
5289 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
5290 https://bugzilla.gnome.org/show_bug.cgi?id=767463
5292 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5295 Automatic update of common submodule
5296 From ac2f647 to f363b32
5298 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5300 * gst/rtsp-server/rtsp-sdp.c:
5301 * gst/rtsp-server/rtsp-sdp.h:
5302 * gst/rtsp-server/rtsp-stream.c:
5303 * gst/rtsp-server/rtsp-stream.h:
5304 sdp: add rollover counters for all sender SSRC
5305 We add different crypto sessions in MIKEY, one for each sender
5306 SSRC. Currently, all of them will have the same security policy, 0.
5307 The rollover counters are obtained from the srtpenc element using the
5309 https://bugzilla.gnome.org/show_bug.cgi?id=730539
5311 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5313 * gst/rtsp-server/rtsp-media-factory.h:
5314 * gst/rtsp-server/rtsp-server.h:
5315 docs: fix some typos
5317 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
5319 * gst/rtsp-server/Makefile.am:
5320 g-i: pass compiler env to g-ir-scanner
5321 It's what introspection.mak does as well. Should
5322 fix spurious build failures on gnome-continuous
5323 (caused by g-ir-scanner getting compiler details
5324 via python which is broken in some environments
5325 so passing the compiler details bypasses that).
5327 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
5329 * gst/rtsp-server/rtsp-session.c:
5330 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
5331 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
5332 https://bugzilla.gnome.org/show_bug.cgi?id=766619
5334 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
5336 * gst/rtsp-sink/gstrtspclientsink.c:
5337 rtspclientsink: Check return value of sscanf
5338 And just make sure we always have 0/0 if we have an error
5341 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
5343 * gst/rtsp-server/rtsp-stream.c:
5344 * tests/check/gst/rtspserver.c:
5345 * tests/check/gst/stream.c:
5346 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
5347 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
5348 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
5349 - Create unit test for shared media.
5350 https://bugzilla.gnome.org/show_bug.cgi?id=764744
5352 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5354 * gst/rtsp-server/rtsp-stream.c:
5355 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
5356 For IPv6 addresses, binding to a multicast group does not work on Linux
5357 either. Always bind to ANY and then later join the multicast group.
5358 https://bugzilla.gnome.org/show_bug.cgi?id=764679
5360 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
5363 Automatic update of common submodule
5364 From 6f2d209 to ac2f647
5366 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
5368 * gst/rtsp-server/rtsp-thread-pool.c:
5369 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
5370 Clarified why it is necessary to add source information to
5371 GstRTSPThreadImpl. See the reported bug in GLib:
5372 https://bugzilla.gnome.org/show_bug.cgi?id=720186
5373 for more information.
5374 https://bugzilla.gnome.org/show_bug.cgi?id=761702
5376 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
5378 * examples/Makefile.am:
5379 examples: Clean up CFLAGS/LDADD even more
5380 The internal .la should come first and is part of LDADD, as is
5383 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5385 * examples/Makefile.am:
5386 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5388 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5390 * gst/rtsp-server/Makefile.am:
5391 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5393 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5395 * gst/rtsp-server/rtsp-client.c:
5396 * gst/rtsp-server/rtsp-media-factory.c:
5397 * gst/rtsp-server/rtsp-media-factory.h:
5398 * gst/rtsp-server/rtsp-media.c:
5399 * gst/rtsp-server/rtsp-media.h:
5400 * gst/rtsp-server/rtsp-sdp.c:
5401 * gst/rtsp-server/rtsp-stream.c:
5402 * gst/rtsp-server/rtsp-stream.h:
5403 rtsp-server: Implement clock signalling according to RFC7273
5404 For NTP and PTP clocks we signal the actual clock that is used and signal
5405 the direct media clock offset.
5406 For all other clocks we at least signal that it's the local sender clock.
5407 This allows receivers to know which clock was used to generate the media and
5408 its RTP timestamps. Receivers can then implement network synchronization,
5409 either absolute or at least relative by getting the sender clock rate directly
5410 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5412 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5414 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5416 * gst/rtsp-sink/gstrtspclientsink.c:
5417 rtspclientsink: Add support for setting the multicast interface
5418 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5420 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5422 * gst/rtsp-server/rtsp-media-factory.c:
5423 * gst/rtsp-server/rtsp-media-factory.h:
5424 * gst/rtsp-server/rtsp-media.c:
5425 * gst/rtsp-server/rtsp-media.h:
5426 * gst/rtsp-server/rtsp-stream.c:
5427 * gst/rtsp-server/rtsp-stream.h:
5428 rtsp-media: Add support for setting the multicast interface
5429 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5431 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5433 * gst/rtsp-sink/gstrtspclientsink.c:
5434 rtspclientsink: use new gst_element_class_add_static_pad_template()
5435 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5437 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5442 === release 1.8.0 ===
5444 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5450 * gst-rtsp-server.doap:
5453 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5455 * gst/rtsp-server/rtsp-stream.c:
5456 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5457 This would get us NO_PREROLL in the bin again and break seeking.
5458 Thanks to Carlos Rafael Giani for helping to debug this!
5459 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5461 === release 1.7.91 ===
5463 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5469 * gst-rtsp-server.doap:
5472 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5474 * gst/rtsp-server/rtsp-stream.c:
5475 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5476 Without this, RECORD pipelines are broken because
5477 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5478 added later. Previously it was there earlier and due to NO_PREROLL caused the
5479 pipeline to preroll immediately
5480 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5481 as the corresponding code previously was only for PLAY pipelines.
5482 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5484 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5486 * gst/rtsp-server/rtsp-stream.c:
5487 rtsp-stream: Fix typo in the docstring
5488 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5490 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5492 * gst/rtsp-server/rtsp-stream.c:
5493 rtsp-stream: Disable multicast loopback for all our sockets
5494 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5495 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5496 loopback setting on the socket... while udpsink does which unfortunately has
5497 no effect here on Windows but on Linux.
5498 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5500 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5502 * tests/check/gst/stream.c:
5503 stream tests: added new tests
5504 Test a case when the address pool only contains multicast addresses
5505 and the client is requesting unicast udp.
5506 Added tests for multicast ports allocation.
5507 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5509 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5511 * gst/rtsp-server/rtsp-stream.c:
5512 rtsp-stream: Only bind multicast sockets to ANY on Windows
5513 On Linux it is still needed to bind to the multicast address
5514 to filter out random other packets, while on Windows binding
5515 to multicast addresses just fails.
5517 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5519 * gst/rtsp-server/rtsp-stream.c:
5520 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5521 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5522 addresses, which is something that used to work before. For unicast addresses
5523 if the pool contains none, we just allocate them as if there is no pool at
5525 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5527 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5529 * gst/rtsp-server/rtsp-client.c:
5530 * gst/rtsp-server/rtsp-stream.c:
5531 rtsp-server: Fix indentation
5533 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5535 * gst/rtsp-server/rtsp-stream.c:
5536 rtsp-stream: Don't bind the sockets to multicast addresses
5537 This works on Linux but fails completely on Windows. You're supposed
5538 to bind to ANY and then join the multicast group.
5539 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5541 === release 1.7.90 ===
5543 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5549 * gst-rtsp-server.doap:
5552 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5555 Automatic update of common submodule
5556 From b64f03f to 6f2d209
5558 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5560 * gst/rtsp-sink/gstrtspclientsink.c:
5561 * tests/check/gst/rtspclientsink.c:
5562 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5563 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5565 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5567 * tests/check/gst/media.c:
5568 * tests/check/gst/rtspclientsink.c:
5569 * tests/check/gst/rtspserver.c:
5570 * tests/check/gst/stream.c:
5571 tests: unit test fixes
5572 Removed port allocation test from the media suite.
5573 The port allocation failure is now in the stream suite.
5575 Make sure that the media is suspended after the DESCRIBE request
5576 before reconfiguring the UDP sinks.
5578 In the RECORD case we have to set async property to false
5579 for the appsink element in the test in order to make sure
5580 that the media pipeline doesn't hang in start_preroll().
5581 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5583 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5585 * gst/rtsp-server/rtsp-client.c:
5586 * gst/rtsp-server/rtsp-stream.c:
5587 * gst/rtsp-server/rtsp-stream.h:
5588 rtsp-stream: postpone UDP socket allocation until SETUP
5589 Postpone the allocation of the UDP sockets until we know
5590 what transport has been chosen by the client.
5591 Both unicast and multicast UDP sources are created in one
5593 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5595 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5597 * gst/rtsp-server/rtsp-stream.c:
5598 rtsp-stream: postpone the creation of the UDP sources
5599 Code refactoring: allocate the UDP ports after the sender and
5600 the reciver parts have been created.
5601 We postpone the creation of the UDP sources until the UDP
5602 ports have been allocated.
5603 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5605 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5607 * gst/rtsp-server/rtsp-stream.c:
5608 rtsp-stream: added function for setting UDP sources to PLAYING state
5609 Code refactoring: Introduced a function for setting UDP sources
5611 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5613 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5615 * gst/rtsp-server/rtsp-stream.c:
5616 rtsp-stream: added function for creating and configuring UDP sources
5617 Code refactoring: create and configure UDP sources in a separate function.
5618 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5620 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5622 * gst/rtsp-server/rtsp-stream.c:
5623 rtsp-stream: added function for RTP/RTCP socket configuration
5624 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5625 in a separate function.
5626 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5628 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5630 * gst/rtsp-server/rtsp-stream.c:
5631 rtsp-stream: added function for creating and configuring UDP sinks
5632 Code refactoring: create and configure UDP sinks in a separate function.
5633 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5635 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5637 * gst/rtsp-server/rtsp-stream.c:
5638 rtsp-stream: added helper function for creating the sender/receiver parts
5639 Code refactoring: introduced helper function for creating
5640 the receiver and the sender parts of the streaming pipeline.
5641 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5643 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5648 === release 1.7.2 ===
5650 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5656 * gst-rtsp-server.doap:
5659 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5661 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5662 uninstalled.pc: add support for non libtool build systems
5663 Currently the .la path is provided which requires to use libtool as
5664 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5665 It is fine as long as the application is built using libtool.
5666 So currently it is not possible to compile a GStreamer application
5667 within gst-uninstalled with CMake or other build system different
5669 This patch allows to do the following in gst-uninstalled env:
5670 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5671 gstreamer-rtsp-server-1.0)
5672 Previously it required to prepend libtool --mode=link
5673 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5675 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5677 * gst/rtsp-sink/gstrtspclientsink.c:
5678 rtspclientsink: remove check for impossible condition
5679 Goto error label checks stream to see if it needs to be unreferenced before
5680 returning, but this goto jumps happens before the stream is ever set, so it
5681 will always be NULL in this error label.
5684 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5686 * gst/rtsp-sink/gstrtspclientsink.c:
5687 rtspclientsink: clean switch statements
5688 Coverity demands for fallthrough statements to be clearly commented,
5689 to distinguish from accidental fall throughs. And it also needs all
5690 cases to finish with a break, even if the break is never going to be
5691 executed like in the case of a continue jump.
5695 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5697 * tests/check/Makefile.am:
5698 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5699 To get the CK_DEFAULT_TIMEOUT defined for all tests
5700 Also removes a 120 seconds timeout that was set as default
5701 explicitly in this module
5702 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5704 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5708 Automatic update of common submodule
5709 From 86e4663 to b64f03f
5711 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5713 * gst/rtsp-server/rtsp-media.c:
5714 rtsp-media: fix state_lock not locked again when preroll fails
5715 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5717 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5720 configure: Move plugin specific flags below all the others
5721 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5722 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5724 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5726 * gst/rtsp-sink/gstrtspclientsink.c:
5727 rtspclientsink: Simplify slightly using new -base API
5728 Use the new Mikey and SDP API in the base plugins libs
5729 to simplify some code.
5730 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5732 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5737 * gst/rtsp-sink/Makefile.am:
5738 * gst/rtsp-sink/gstrtspclientsink.c:
5739 * gst/rtsp-sink/gstrtspclientsink.h:
5740 * gst/rtsp-sink/plugin.c:
5741 * tests/check/Makefile.am:
5742 * tests/check/gst/rtspclientsink.c:
5743 rtspsink: Add rtspclientsink element
5744 Add an rtspclientsink element that accepts streams for which
5745 there is a registered payloader and sends them to
5746 an RTSP server using RECORD.
5747 Sending is synchronised to the pipeline clock. Payload-types
5748 are automatically selected. The 'new-payloader' signal is fired
5749 for custom configuration of payloaders when they are created.
5750 Can now stream a movie like this:
5752 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5753 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5755 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5756 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5757 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5759 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5761 * gst/rtsp-server/rtsp-stream.c:
5762 * gst/rtsp-server/rtsp-stream.h:
5763 rtsp-stream: Add functions for using rtsp-stream from the client
5764 Add a boolean to indicate that the rtsp-stream is running on the
5765 'client' side of an RTSP connection, for sending streams via
5766 RECORD. In that case, the roles of the client/server ports
5767 in transport setup are swapped.
5768 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5770 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5772 * gst/rtsp-server/rtsp-sdp.c:
5773 * gst/rtsp-server/rtsp-sdp.h:
5774 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5775 A new function that adds info from a GstRTSPStream into an SDP message.
5776 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5778 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5780 * gst/rtsp-server/rtsp-media.c:
5781 rtsp-media: Fix mutex beeing unlocked while they should be locked
5782 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5784 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5786 * gst/rtsp-server/rtsp-media-factory.c:
5787 rtsp-media-factory: add missing break in "clock" property setter
5790 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5792 * gst/rtsp-server/rtsp-stream.c:
5793 rtsp-stream: fixed assert during update transport
5794 When RTSP server trying update transport during multicast, it throws an
5795 assert. The assert is thrown because it is trying to get the parent of
5796 an non-existing funnel element.
5797 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5799 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5801 * gst/rtsp-server/rtsp-permissions.h:
5802 * gst/rtsp-server/rtsp-thread-pool.h:
5803 * gst/rtsp-server/rtsp-token.h:
5804 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5805 gtk-doc can handle static inline functions just fine these days,
5806 there's no need for this stuff any more.
5808 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5810 * gst/rtsp-server/rtsp-media.c:
5811 * gst/rtsp-server/rtsp-sdp.c:
5812 sdp: replace duplicated codes to call new base sdp apis
5813 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5815 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5817 * examples/test-netclock.c:
5818 test-netclock: Use the new API to configure a clock directly
5820 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5822 * gst/rtsp-server/rtsp-media-factory.c:
5823 * gst/rtsp-server/rtsp-media-factory.h:
5824 * gst/rtsp-server/rtsp-media.c:
5825 * gst/rtsp-server/rtsp-media.h:
5826 rtsp-media: Add API to directly configure a clock on the media pipelines
5828 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5830 * gst/rtsp-server/rtsp-media.c:
5831 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5833 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5835 * gst/rtsp-server/rtsp-media-factory.c:
5836 rtsp-media-factory: Add FIXME for 2.0
5838 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5840 * gst/rtsp-server/rtsp-stream.c:
5841 rtsp-stream: Fix indentation
5843 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5845 * gst/rtsp-server/rtsp-media.c:
5846 rtsp-media: Do not prepare media after media times out
5847 Deferred calls to start_prepare() can be deferred past the point until
5848 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5849 prepared to wait. Previously there was no lock and no check for this
5850 situation. This meant that a media could be prepared and unprepared
5851 simultaneously by two different threads. Now a lock is in place and a
5852 suitable check is done.
5853 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5855 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5857 * gst/rtsp-server/rtsp-client.c:
5858 * gst/rtsp-server/rtsp-media-factory.c:
5859 * gst/rtsp-server/rtsp-media-factory.h:
5860 * gst/rtsp-server/rtsp-media.c:
5861 * gst/rtsp-server/rtsp-media.h:
5862 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5863 Without TEARDOWN it might be desireable to keep the media running and continue
5864 sending data to the client, even if the RTSP connection itself is
5866 Only do this for session medias that have only UDP transports. If there's at
5867 least on TCP transport, it will stop working and cause problems when the
5868 connection is disconnected.
5869 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5871 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5876 === release 1.7.1 ===
5878 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5884 * gst-rtsp-server.doap:
5887 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5890 configure: Make -Bsymbolic check work with clang.
5891 Update the -Bsymbolic check with the version glib has. This version
5893 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5895 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5897 * gst/rtsp-server/rtsp-session-pool.c:
5898 rtsp-session-pool: Avoid dollar sign ($) in session ids
5899 Live555 in VLC strips off dollar signs and then gets very confused,
5900 we don't loose too much entropy by just skipping it.
5902 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5904 * gst/rtsp-server/rtsp-address-pool.h:
5905 * gst/rtsp-server/rtsp-auth.h:
5906 * gst/rtsp-server/rtsp-client.h:
5907 * gst/rtsp-server/rtsp-media-factory-uri.h:
5908 * gst/rtsp-server/rtsp-media-factory.h:
5909 * gst/rtsp-server/rtsp-media.h:
5910 * gst/rtsp-server/rtsp-mount-points.h:
5911 * gst/rtsp-server/rtsp-permissions.h:
5912 * gst/rtsp-server/rtsp-server.h:
5913 * gst/rtsp-server/rtsp-session-media.h:
5914 * gst/rtsp-server/rtsp-session-pool.h:
5915 * gst/rtsp-server/rtsp-session.h:
5916 * gst/rtsp-server/rtsp-stream-transport.h:
5917 * gst/rtsp-server/rtsp-stream.h:
5918 * gst/rtsp-server/rtsp-thread-pool.h:
5919 * gst/rtsp-server/rtsp-token.h:
5920 rtsp-server: Add g_autoptr() support to all types
5921 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5923 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5925 * gst/rtsp-server/rtsp-stream.c:
5926 rtsp-stream: fixed valgrind error
5927 Fixed the valgrind error in unit test. The UDP source created during
5928 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5930 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5932 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5936 Automatic update of common submodule
5937 From b319909 to 86e4663
5939 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5941 * gst/rtsp-server/rtsp-client.c:
5942 rtsp-client: suspend media during setup request
5943 SETUP request from clients needs to suspend the media to clear the
5944 prerolled buffers. Otherwise it will not affect the prerolled buffer
5945 and the prerolled buffers will be incorrect (for example block-size
5946 from setup request will not affect the prerolled buffer unless the
5947 media is suspended).
5948 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5950 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5952 * gst/rtsp-server/rtsp-stream.c:
5953 rtsp-stream: create stream pipeline based on transport
5954 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5955 only UDP is set as the transport protocol, it will not add the extra tee
5956 or queue element to the pipeline. Both these elements will be added, if
5957 it supports both TCP and UDP protocols. This improves the pipeline
5958 performance when one protocol is present.
5959 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5961 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5963 * gst/rtsp-server/rtsp-stream.c:
5964 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5965 Adding them when not needed will start some logic inside rtpbin that might be
5966 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5967 would start up a rtpjitterbuffer and behave in weird ways.
5968 We still set up the UDP sources for RTP receiving for a sender media to be
5969 able to receive any packets sent by the client for NAT traversal. They will
5970 all go to a fakesink though.
5971 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5972 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5973 receive ASYNC_DONE after a seek.
5974 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5976 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5978 * gst/rtsp-server/rtsp-stream.c:
5979 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5980 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5981 Previously we were only setting this for sender sockets, which caused looped
5982 back packets to be received on Windows if a multicast transport was used.
5984 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5986 * examples/test-record-auth.c:
5987 * examples/test-record.c:
5988 examples: Actually use the provided port in the record examples
5990 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5992 * examples/test-record-auth.c:
5993 test-record-auth: Add the option to build in TLS support
5995 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5997 * examples/test-auth.c:
5998 test-auth: Use an 'anonymous' user for unauthenticated default
5999 There's a comment on one of the resources that 'user' and 'admin'
6000 shouldn't even be able to see it, but they can if the default
6001 token is 'admin2', since that gives them access anyway.
6003 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
6005 * examples/.gitignore:
6006 * examples/Makefile.am:
6007 * examples/test-record-auth.c:
6008 Add test-record-auth example
6010 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
6012 * gst/rtsp-server/rtsp-client.c:
6013 * tests/check/gst/client.c:
6014 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
6016 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
6018 * gst/rtsp-server/rtsp-server.c:
6019 rtsp-server: Change the logic so we don't pop a NULL context
6020 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
6021 will sometimes fail. This call is made before any context is pushed
6022 resulting in an attempt to pop a NULL context.
6023 https://bugzilla.gnome.org/show_bug.cgi?id=757949
6025 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
6027 * tests/check/gst/rtspserver.c:
6028 rtspserver: Add udp-mcast transport SETUP test
6029 Refactor utility functions in the test file so they can handle
6030 more than UDP and TCP as lower transport.
6031 https://bugzilla.gnome.org/show_bug.cgi?id=756969
6033 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
6035 * gst/rtsp-server/rtsp-stream.c:
6036 rtsp-stream: Always unref return value of gst_object_get_parent()
6037 Fixes a leak of a GstBin in the udp-mcast case.
6038 https://bugzilla.gnome.org/show_bug.cgi?id=756968
6040 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
6043 Automatic update of common submodule
6044 From b99800a to b319909
6046 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
6049 Use new GST_ENABLE_EXTRA_CHECKS #define
6050 https://bugzilla.gnome.org/show_bug.cgi?id=756870
6052 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6055 Automatic update of common submodule
6056 From 6babecd to b99800a
6058 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6061 Update GLib dependency to 2.40.0
6063 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6065 * examples/test-mp4.c:
6066 * gst/rtsp-server/rtsp-stream.c:
6067 stream: listen to sender ssrc signals
6068 https://bugzilla.gnome.org/show_bug.cgi?id=746747
6070 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
6073 common: update for new suppression
6074 Makes check-valgrind pass with glib 2.46
6076 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6078 * gst/rtsp-server/rtsp-media.c:
6079 rtsp-media: Take reference to media that will be prepared
6080 default_prepare() takes a transfer-none reference GstRTSPMedia object.
6081 Later on a g_idle_source_new() is created and a pointer to the media
6082 object is passed as user data. If the media is freed before the idle
6083 source is dispatched the media object pointer is invalid, but the idle
6084 source callback expects it to still be valid. To fix this a reference to
6085 the media object is taken when registering the source callback function
6086 and a corresponding release of the reference is done when the souce is
6088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
6090 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
6092 * examples/test-launch.c:
6093 * examples/test-mp4.c:
6094 * examples/test-ogg.c:
6095 * examples/test-record.c:
6096 * examples/test-uri.c:
6097 rtsp-server: Fix memory leaks when context parse fails
6098 When g_option_context_parse fails, context and error variables are not getting free'd
6099 which results in memory leaks. Free'ing the same.
6100 And replacing g_error_free with g_clear_error, which checks if the error being passed
6101 is not NULL and sets the variable to NULL on free'ing.
6102 https://bugzilla.gnome.org/show_bug.cgi?id=753863
6104 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
6109 === release 1.6.0 ===
6111 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6117 * gst-rtsp-server.doap:
6120 === release 1.5.91 ===
6122 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
6128 * gst-rtsp-server.doap:
6131 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
6133 * docs/libs/gst-rtsp-server-sections.txt:
6134 * gst/rtsp-server/rtsp-stream.c:
6135 stream: fix docs for recently-added get/set_buffer_size API
6136 https://bugzilla.gnome.org/show_bug.cgi?id=749095
6138 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
6140 * gst/rtsp-server/rtsp-media.c:
6141 rtsp-media: Don't crash on encrypted RTX SDP
6142 In parse_keymgmt(), don't mutate the input string that's been passed
6143 as const, especially since we might need the original value again if
6144 the same key info applies to multiple streams (RTX, for example).
6145 https://bugzilla.gnome.org/show_bug.cgi?id=754753
6147 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
6149 * examples/test-mp4.c:
6150 test-mp4: Support filenames with spaces in them. Error out on too few arguments
6152 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
6154 * examples/test-record.c:
6155 test-record: Check parameter count and print out help
6156 If no launch pipeline was supplied, print out some help
6158 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
6160 * gst/rtsp-server/rtsp-media.c:
6161 * gst/rtsp-server/rtsp-stream.c:
6162 * gst/rtsp-server/rtsp-stream.h:
6163 rtsp-stream: Implement UDP buffer size setting.
6164 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
6166 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
6167 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
6169 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
6171 * gst/rtsp-server/rtsp-media.h:
6172 rtsp-media: Fix small typo causing gtk-doc to complain
6174 === release 1.5.90 ===
6176 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
6182 * gst-rtsp-server.doap:
6185 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6187 * gst/rtsp-server/rtsp-media-factory.c:
6188 media-factory: get port number through gst_rtsp_url_get_port
6189 https://bugzilla.gnome.org/show_bug.cgi?id=753473
6191 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
6193 * tests/check/gst/media.c:
6194 media-test: Removing unnecessary assertion
6195 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6197 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6199 * gst/rtsp-server/rtsp-server.c:
6200 Document that source keeps a ref on server until it's destroyed
6201 https://bugzilla.gnome.org/show_bug.cgi?id=749227
6203 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6205 * tests/check/gst/media.c:
6206 media-test: Test for multiple dynamic payload
6207 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6209 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6211 * gst/rtsp-server/rtsp-media.c:
6212 media: Only add fakesink once per pipeline
6213 The intention is to prevent going PLAYING state before pads are created.
6214 If there was mutilple dynamic payload, it would leak few fakesink and
6215 actually prevent from ever reaching playing state.
6216 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6218 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6220 * gst/rtsp-server/rtsp-media.c:
6221 Revert "rtsp-media: Only add 1 fakesink per pipeline"
6222 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
6224 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6226 * gst/rtsp-server/rtsp-media.c:
6227 rtsp-media: Only add 1 fakesink per pipeline
6228 There should be only one fakesink per pipeline, not per dynpay. This
6229 would lead to element naming clash.
6231 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
6233 * gst/rtsp-server/rtsp-media.c:
6234 rtsp-media: assertion error due to wrong condition check
6235 In media to caps function, reserved_keys array is being used for variable i,
6236 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
6237 changed it to variable j
6238 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6240 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
6242 * gst/rtsp-server/rtsp-media.c:
6243 rtsp-media: Strip keys from the fmtp that we use internally in our caps
6244 Skip keys from the fmtp, which we already use ourselves for the
6245 caps. Some software is adding random things like clock-rate into
6246 the fmtp, and we would otherwise here set a string-typed clock-rate
6247 in the caps... and thus fail to create valid RTP caps
6248 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6250 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6252 * gst/rtsp-server/rtsp-thread-pool.c:
6253 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
6254 https://bugzilla.gnome.org/show_bug.cgi?id=752640
6256 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
6259 Automatic update of common submodule
6260 From f74b2df to 9aed1d7
6262 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
6267 === release 1.5.2 ===
6269 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
6275 * gst-rtsp-server.doap:
6278 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
6280 * gst/rtsp-server/rtsp-client.c:
6281 * gst/rtsp-server/rtsp-client.h:
6282 * tests/check/gst/client.c:
6283 rtsp-client: allow application to decide what requirements are supported
6284 Add "check-requirements" signal and vfunc to allow application
6285 (and subclasses) to check the requirements.
6286 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
6287 https://bugzilla.gnome.org/show_bug.cgi?id=749417
6289 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6292 Automatic update of common submodule
6293 From 6015d26 to f74b2df
6295 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6297 * gst/rtsp-server/rtsp-media.c:
6298 rtsp-media: Always use real payloader when creating streams
6299 A bin that contains the real payloader might be used as payloader. In this
6300 case we have to get the real payloader for the various properties it provides.
6301 Example use cases for this are bins that payload some media and then have
6302 additional elements that add metadata or RTP extension headers to the stream.
6303 https://bugzilla.gnome.org/show_bug.cgi?id=750800
6305 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
6307 * examples/test-netclock-client.c:
6308 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
6310 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
6312 * examples/test-netclock-client.c:
6313 * examples/test-netclock.c:
6314 test-netclock: Use new ntp-time-source property on rtpbin
6315 Select the clock time to be used as NTP time source. This allows proper
6316 synchronization between receivers, independent of sharing base times, and just
6317 requires them to use the same clock.
6319 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6321 * examples/test-netclock-client.c:
6322 * examples/test-netclock.c:
6323 test-netclock: Setting the same base time on sender and receiver is not necessary
6324 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
6326 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6328 * gst/rtsp-server/rtsp-stream.c:
6329 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
6330 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6332 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6334 * docs/libs/gst-rtsp-server.types:
6335 docs: add missing types
6336 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6338 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6340 * docs/libs/gst-rtsp-server-sections.txt:
6341 docs: add missing apis
6342 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6344 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
6346 * examples/test-netclock-client.c:
6347 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
6349 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6351 * docs/libs/gst-rtsp-server-sections.txt:
6352 * gst/rtsp-server/rtsp-auth.c:
6353 * gst/rtsp-server/rtsp-auth.h:
6354 GstRTSPAuth: Add client certificate authentication support
6355 https://bugzilla.gnome.org/show_bug.cgi?id=750471
6357 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
6359 * examples/test-netclock-client.c:
6360 test-netclock-client: Use new GstClock API to wait for clock synchronization
6362 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
6364 * examples/test-netclock-client.c:
6365 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
6366 A mainloop is needed to get glimagesink to display something on OSX, and
6367 the source-setup signal just makes things a little bit easier.
6369 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
6372 Automatic update of common submodule
6373 From d9a3353 to 6015d26
6375 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
6378 Automatic update of common submodule
6379 From d37af32 to d9a3353
6381 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6384 Automatic update of common submodule
6385 From 21ba2e5 to d37af32
6387 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6390 Automatic update of common submodule
6391 From c408583 to 21ba2e5
6393 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6395 * docs/libs/Makefile.am:
6396 docs: remove variables that we define in the snippet from common
6397 This is syncing our Makefile.am with upstream gtkdoc.
6399 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6402 Automatic update of common submodule
6403 From 44a3517 to c408583
6405 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6410 === release 1.5.1 ===
6412 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6418 * gst-rtsp-server.doap:
6421 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6423 * gst/rtsp-server/rtsp-client.c:
6424 rtsp-client: No flush during Teardown.
6425 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6426 backlog is empty it can happen that just a part of a message will be
6427 sent and rest is in backlog queue. If then flush during teardown
6428 just a part of message will be sent.This can lead to client miss
6429 teardown response since it expect to get the last part of message.
6430 The flushing during teardown was introduced to fix a deadlock that now
6431 is fixed more generally in handle_request by temporary setting backlog
6433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6435 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6437 * tests/check/Makefile.am:
6438 tests: Use AM_TESTS_ENVIRONMENT
6439 Needed by the new automake test runner and the
6440 current version of the common submodule.
6442 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6444 * gst/rtsp-server/rtsp-media.h:
6445 * gst/rtsp-server/rtsp-stream.h:
6446 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6448 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6450 * gst/rtsp-server/rtsp-media.c:
6451 rtsp-media: Mark some more functions static
6453 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6455 * gst/rtsp-server/rtsp-media.c:
6456 rtsp-media: Only unblock the media in suspend() when actually changing the state
6457 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6459 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6461 * examples/test-video-rtx.c:
6462 examples: Use AVPF profile for the RTX example
6464 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6466 * gst/rtsp-server/rtsp-sdp.c:
6467 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6469 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6471 * gst/rtsp-server/rtsp-stream.c:
6472 rtsp-stream: get valid clock-rate from last-sample
6473 clock-rate in last-sample's caps is integer, not unsigned.
6474 To get this value properly, variable needs to be type-casted to int.
6475 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6477 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6481 autogen.sh: only run autopoint if gettext requested in configure.ac
6482 Not just because there happens to be a po directory.
6483 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6485 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6488 Revert "configure.ac: uncomment gettext version setup"
6489 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6490 We don't need a gettext setup here and there's no po
6491 directory either, so no reason why autopoint would be
6492 run in the first place.
6493 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6495 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6497 * examples/test-multicast.c:
6498 * examples/test-multicast2.c:
6499 * examples/test-sdp.c:
6500 * examples/test-video-rtx.c:
6501 * examples/test-video.c:
6502 * tests/test-cleanup.c:
6503 * tests/test-reuse.c:
6504 Fix timeout function signatures across tests and examples
6506 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6508 * tests/check/Makefile.am:
6509 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6510 Make sure the test environment is set up.
6511 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6513 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6516 configure: bump automake requirement to 1.14 and autoconf to 2.69
6517 This is only required for builds from git, people can still
6518 build tarballs if they only have older autotools.
6519 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6521 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6524 configure.ac: uncomment gettext version setup
6525 Fixes autogen.sh. It would run autopoint, which would complain
6526 that it could not find the gettext version in configure.ac.
6527 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6529 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6531 * examples/test-video-rtx.c:
6532 test-video-rtx: set exact payload type to PCMA payloader
6533 Setting wrong payload type causes failure to do retransmission through audio stream
6534 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6536 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6538 * gst/rtsp-server/rtsp-media.c:
6539 * gst/rtsp-server/rtsp-stream.c:
6540 * gst/rtsp-server/rtsp-stream.h:
6541 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6542 Because of duplicated g_signal_connect for request-aux-sender signal,
6543 wrong stream pointer is passed to the signal handler.
6544 Instead of passing each stream, pass stream array and get the relevant stream.
6545 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6547 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6551 Update autogen.sh to latest version from common
6552 Fixes build after aclocal_check etc. helpers have been removed.
6554 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6557 Automatic update of common submodule
6558 From bc76a8b to c8fb372
6560 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6562 * gst/rtsp-server/rtsp-stream.c:
6563 rtsp-stream: Limit the queues to 1 buffer
6564 We only need them to be able to pre-roll, queueing up more data here
6565 is only going to harm latency and memory usage.
6567 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6569 * gst/rtsp-server/rtsp-stream.c:
6570 rtsp-stream: Update comment and ASCII art to the latest code
6571 We have a queue in front of the udpsink too to prevent the pipeline from
6574 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6576 * gst/rtsp-server/rtsp-stream.c:
6577 rtsp-media: Properly return first rtptime
6578 Instead we where returning first GstBuffer timestamp. This would result
6579 in clock skew and unwanted behaviour in RTSP playback.
6580 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6582 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6584 * gst/rtsp-server/rtsp-stream.c:
6585 rtsp-stream: Don't leave buffer mapped
6586 If the seq is NULL, the RTP buffer was left mapped. We should always
6589 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6594 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6596 * gst/rtsp-server/rtsp-media-factory.c:
6597 * tests/check/gst/client.c:
6598 Fix double semicolons
6600 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6602 * gst/rtsp-server/rtsp-stream.c:
6603 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6604 This gives more accurate values than asking the payloader. There might be
6605 queueing happening between the payloader and the sink.
6606 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6608 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6610 * gst/rtsp-server/rtsp-media.c:
6611 rtsp-media: Don't seek for PLAY if the position will not change
6612 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6614 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6616 * gst/rtsp-server/rtsp-media.c:
6617 rtsp-media: Don't include payload type in the caps for framesize
6618 When the sdp media attribute framesize are converted to caps
6619 the <payload> should not be included.
6620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6621 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6623 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6625 * gst/rtsp-server/rtsp-sdp.c:
6626 rtsp-sdp: add payload type to the sdp framesize attribute
6627 The sdp framesize attribute is desribed in RFC6064. It is specified
6628 for payloading of H263 and has the following form
6629 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6630 should be added to the caps in a payloader and the <payload type> should
6631 be added by the rtsp-server.
6632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6634 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6636 * examples/test-uri.c:
6637 examples: test-uri: fix tainted variable
6638 Insignificant but this keeps Coverity happy.
6641 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6643 * examples/.gitignore:
6644 * examples/Makefile.am:
6645 * examples/test-netclock-client.c:
6646 * examples/test-netclock.c:
6647 examples: Add a simple example of network synch for live streams.
6648 An example server and client that works for synchronising live streams
6649 only - as it can't support pause/play.
6651 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6653 * gst/rtsp-server/rtsp-media-factory.c:
6654 * gst/rtsp-server/rtsp-media-factory.h:
6655 rtsp-media-factory: Add functions to set/get the media gtype
6656 Allow specifying the GType of a GstRtspMedia subclass to create
6657 as a simpler way to get the factory to create a custom
6658 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6660 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6662 * gst/rtsp-server/rtsp-media.c:
6663 rtsp-media: fix double unlock in _get_buffer_size()
6664 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6665 because of double g_mutex_unlock () usage.
6666 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6668 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6670 * gst/rtsp-server/rtsp-session-pool.c:
6671 * gst/rtsp-server/rtsp-session.c:
6672 * gst/rtsp-server/rtsp-session.h:
6673 rtsp-session: Use monotonic time for RTSP session timeout
6674 Changed RTSP session timeout handling to monotonic time
6675 and deprecating the API for current system time.
6676 This fixes timeouts when the system time changes.
6677 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6679 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6681 * gst/rtsp-server/rtsp-client.c:
6682 * gst/rtsp-server/rtsp-media.c:
6683 rtsp-client: Only error out in PLAY if seeking actually failed
6684 If the media was just not seekable, we continue from whatever position we are
6685 and let the client decide if that is what is wanted or not.
6686 Only if the actual seek failed, we can't really recover and should error out.
6688 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6690 * gst/rtsp-server/rtsp-stream.c:
6691 rtsp-stream: Add necessary queues between tee and multiudpsink
6692 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6694 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6696 * gst/rtsp-server/rtsp-client.c:
6697 * gst/rtsp-server/rtsp-media.c:
6698 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6699 Instead error out properly the same way as if the SEEKING query already
6702 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6704 * gst/rtsp-server/rtsp-stream.h:
6705 rtsp-stream: minor code formatting fix
6707 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6709 * gst/rtsp-server/rtsp-media.c:
6710 rtsp-media: fix logic for collect_streams
6711 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6712 all streams it knows if it got any, and can check if the transport mode is OK.
6715 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6717 * gst/rtsp-server/rtsp-media.c:
6718 rtsp-media: Don't set the transport mode based on what elements we find
6719 Just print a warning if the one that was set before disagrees with what
6720 elements we found. It must already be set to something before as this
6721 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6722 and we would reject ANNOUNCE if the RECORD flag was not set.
6724 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6726 * tests/check/gst/rtspserver.c:
6727 tests: rtspserver: rename shadowed variable
6728 We have two different 'sink' variables here,
6729 rename one of them for clarity.
6731 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6733 * gst/rtsp-server/rtsp-client.c:
6734 rtsp-client: fix awkward if clause
6736 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6738 * examples/test-uri.c:
6739 examples: test-uri: improve uri argument handling and accept file names
6740 Print an error if the argument passed is not a URI and can't
6741 be converted into one, or no arguments have been provided.
6743 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6745 * examples/test-uri.c:
6746 examples: test-uri: don't remove mount point after 10 seconds
6747 It's very irritating when trying to test stuff repeatedly
6748 and serves no real purpose other than showing that it can
6751 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6753 * examples/.gitignore:
6754 examples: add new test-record to .gitignore
6756 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6758 * examples/test-record.c:
6759 * gst/rtsp-server/rtsp-client.c:
6760 * gst/rtsp-server/rtsp-media-factory.c:
6761 * gst/rtsp-server/rtsp-media-factory.h:
6762 * gst/rtsp-server/rtsp-media.c:
6763 * gst/rtsp-server/rtsp-media.h:
6764 * tests/check/gst/rtspserver.c:
6765 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6767 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6769 * examples/test-record.c:
6770 test-record: Set latency for playback-style example to 2s instead of 200ms
6772 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6774 * tests/check/gst/rtspserver.c:
6775 tests: add some unit tests for ANNOUNCE and RECORD
6776 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6778 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6780 * gst/rtsp-server/rtsp-client.c:
6781 rtsp-client: fix a couple of leaks in handle_announce
6783 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6785 * gst/rtsp-server/rtsp-media-factory.c:
6786 * gst/rtsp-server/rtsp-media-factory.h:
6787 * gst/rtsp-server/rtsp-media.c:
6788 * gst/rtsp-server/rtsp-media.h:
6789 rtsp-media: Expose latency setting for setting the rtpbin latency
6791 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6793 * examples/test-record.c:
6794 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6796 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6798 * gst/rtsp-server/rtsp-stream.c:
6799 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6801 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6803 * examples/Makefile.am:
6804 * examples/test-record.c:
6805 * gst/rtsp-server/rtsp-client.c:
6806 * gst/rtsp-server/rtsp-client.h:
6807 * gst/rtsp-server/rtsp-media-factory.c:
6808 * gst/rtsp-server/rtsp-media-factory.h:
6809 * gst/rtsp-server/rtsp-media.c:
6810 * gst/rtsp-server/rtsp-media.h:
6811 * gst/rtsp-server/rtsp-session-media.c:
6812 * gst/rtsp-server/rtsp-stream.c:
6813 * gst/rtsp-server/rtsp-stream.h:
6814 Add initial support for RECORD
6815 We currently only support media that is RECORD or PLAY only, not both at once.
6816 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6818 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6820 * gst/rtsp-server/rtsp-stream.c:
6821 rtsp-stream: RTCP and RTP transport cache cookies seperated
6822 RTCP packets were not sent because the same tr_cache_cookie was used for
6823 both RTP and RTCP. So only one of the tr_cache lists were populated
6824 depending on which one was sent first. If the tr_cache list is not
6825 populated then no packets can be sent. Most often this happened to be
6826 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6827 resulted in both the tr_cache_lists to be populated regardless of which
6829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6831 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6833 * gst/rtsp-server/rtsp-stream.c:
6834 rtsp-stream: fix false compiler warning
6835 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6837 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6839 * gst/rtsp-server/rtsp-client.c:
6840 rtsp-client: log interleaved data received
6842 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6844 * gst/rtsp-server/rtsp-client.c:
6845 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6847 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6849 * gst/rtsp-server/rtsp-client.c:
6850 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6852 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6854 * gst/rtsp-server/rtsp-client.c:
6855 rtsp-client: Use a random session ID in the SDP
6856 RFC4566 Section 5.2 says that it should make the username, session id,
6857 nettype, addrtype and unicast address tuple globally unique. Always using
6858 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6859 Instead let's create a 64 bit random number, which at least brings us
6860 closer to the goal of global uniqueness.
6861 https://tools.ietf.org/html/rfc4566#section-5.2
6863 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6865 * examples/test-launch.c:
6866 * examples/test-mp4.c:
6867 * examples/test-ogg.c:
6868 * examples/test-uri.c:
6869 examples: Don't call gst_init() and gst_get_option_group()
6870 The latter calls the former at the appropriate time.
6872 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6874 * gst/rtsp-server/rtsp-client.c:
6875 rtsp-client: Drop trailing \0 of RTSP DATA messages
6876 We add a trailing \0 in GstRTSPConnection to make parsing of
6877 string message bodies easier (e.g. the SDP from DESCRIBE) but
6878 for actual data this means we have to drop it or otherwise
6879 create invalid data.
6881 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6883 * gst/rtsp-server/rtsp-stream.c:
6884 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6885 Fixes crash when two threads access handle_new_sample() at the same
6886 time, one for RTP, one for RTCP.
6887 Otherwise, when iterating over the transports cache, it might be modified by
6888 another thread at the same time if the transports cookie has changed.
6889 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6891 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6893 * gst/rtsp-server/rtsp-stream.c:
6894 rtsp-stream: Set format=TIME on our app sources for TCP
6896 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6898 * gst/rtsp-server/rtsp-session-pool.c:
6899 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6900 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6901 RFC 2326 states that session IDs may consist of alphanumeric as well as
6902 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6903 Previously the session ID was URI-escaped, this meant that any character
6904 which was not alphanumeric or any of the characters +-._~ would be
6905 percent encoded. While the RFC (surprisingly) mentions that linear white
6906 space in session IDs should be URI-escaped, it does not say anything
6907 about other characters. Moreover no white space is allowed in the
6908 session ID. Finally the percent character which is the result of
6909 URI-escaping is not allowed in a session ID.
6910 So there is no reason to do any URI-escaping, and now it is removed.
6911 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6913 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6916 Automatic update of common submodule
6917 From f2c6b95 to bc76a8b
6919 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6922 Fix 'make check' from top-level directory
6924 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6926 * examples/test-launch.c:
6927 * examples/test-mp4.c:
6928 * examples/test-ogg.c:
6929 * examples/test-uri.c:
6930 examples: Add command-line parsing and take a 'port' argument
6931 This allows users to run multiple servers on different ports for testing.
6932 Only done for examples that actually take arguments and hence are capable of
6933 outputting different streams for each instance on each port.
6934 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6936 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6938 * gst/rtsp-server/rtsp-client.c:
6939 * gst/rtsp-server/rtsp-client.h:
6940 rtsp-client: Add a send_message default signal handler
6941 This allows subclasses to easily hook into the response sending
6942 mechanism without doing everything from a signal, which seems
6943 awkward from subclasses.
6945 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6948 Automatic update of common submodule
6949 From ef1ffdc to f2c6b95
6951 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6955 configure: add --disable-examples switch
6956 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6958 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6960 * examples/.gitignore:
6961 * examples/Makefile.am:
6962 * examples/test-video-rtx.c:
6963 examples: add a retransmisison example implementing RFC4588
6964 Currently only SSRC-multiplexed rtx streams are supported
6966 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6968 * gst/rtsp-server/rtsp-stream.c:
6969 rtsp-stream: Fix some minor memory leaks
6971 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6973 * gst/rtsp-server/rtsp-media.c:
6974 rtsp-media: Some minor cleanup
6976 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6978 * gst/rtsp-server/rtsp-stream.c:
6979 rtsp-stream: Fix compiler warnings
6980 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6981 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6983 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6984 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6987 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6989 * docs/libs/gst-rtsp-server-sections.txt:
6990 * gst/rtsp-server/rtsp-media-factory.c:
6991 * gst/rtsp-server/rtsp-media-factory.h:
6992 * gst/rtsp-server/rtsp-media.c:
6993 * gst/rtsp-server/rtsp-media.h:
6994 * gst/rtsp-server/rtsp-sdp.c:
6995 * gst/rtsp-server/rtsp-stream.c:
6996 * gst/rtsp-server/rtsp-stream.h:
6997 media: implement ssrc-multiplexed retransmission support
6998 based off RFC 4588 and the server-rtpaux example in -good
7000 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
7002 * gst/rtsp-server/rtsp-client.c:
7003 * gst/rtsp-server/rtsp-stream-transport.c:
7004 * gst/rtsp-server/rtsp-stream.c:
7005 rtsp: Ref transports in hash table.
7006 Also ref streams for transports.
7007 This solves a crash when reciving a rtcp after teardown but before
7009 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
7011 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
7014 Automatic update of common submodule
7015 From 7bb2bce to ef1ffdc
7017 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
7019 * gst/rtsp-server/rtsp-client.c:
7020 client: refactor cleanup of cached media
7022 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
7024 * tests/check/gst/client.c:
7026 The session leak is now fixed, lets remove those FIXME comments.
7028 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
7030 * tests/check/gst/rtspserver.c:
7031 tests: Test to setup two sessions on one connection
7032 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7034 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
7036 * tests/check/gst/rtspserver.c:
7037 tests: Test setup with tcp transport
7038 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7040 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
7042 * gst/rtsp-server/rtsp-client.c:
7043 client: Configure transport after creating session media
7044 The default implementation of configure_client_transport() in
7045 rtsp-client uses the session media when it chooses channels for
7046 interleaved traffic.
7047 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7049 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
7051 * gst/rtsp-server/rtsp-client.c:
7052 * gst/rtsp-server/rtsp-session-media.c:
7053 client: Stop caching media in client when doing setup
7054 If the media has been managed by a session media, it should not be
7055 cached in the client any longer. The GstRTSPSessionMedia object is now
7056 responsible for unpreparing the GstRTSPMedia object using
7057 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
7059 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7061 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7063 * gst/rtsp-server/rtsp-stream.c:
7064 rtsp-stream: unref srtp decoder when leaving bin
7065 https://bugzilla.gnome.org/show_bug.cgi?id=739481
7067 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7069 * gst/rtsp-server/rtsp-client.c:
7070 rtsp-client: mikey memory leaks
7071 https://bugzilla.gnome.org/show_bug.cgi?id=739383
7073 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
7076 Automatic update of common submodule
7077 From 84d06cd to 7bb2bce
7079 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
7082 Parallelise 'make check-valgrind'
7084 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
7087 Automatic update of common submodule
7088 From a8c8939 to 84d06cd
7090 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
7093 Automatic update of common submodule
7094 From 36388a1 to a8c8939
7096 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7098 * gst/rtsp-server/rtsp-media.c:
7099 rtsp-media: deactivate media when shutting down from paused
7100 This was only done when going directly from playing.
7101 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
7103 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7105 * gst/rtsp-server/rtsp-client.c:
7106 * gst/rtsp-server/rtsp-context.h:
7107 rtsp-client: add stream transport to context
7108 We add the stream transport to the context so we can get the configured
7109 client stream transport in the setup request signal.
7110 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
7112 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7114 * gst/rtsp-server/rtsp-stream.c:
7115 stream: release lock even not all transports have been removed
7116 We don't want to keep the lock even we return FALSE because not all the
7117 transports have been removed. This could lead into a deadlock.
7118 https://bugzilla.gnome.org/show_bug.cgi?id=737797
7120 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
7122 * gst/rtsp-server/rtsp-sdp.c:
7123 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
7124 These were renamed in GstRTPBasePayload in 1.0
7126 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7128 * gst/rtsp-server/rtsp-client.c:
7129 client: set session media to NULL without the lock
7130 We need to set session medias to NULL without the client lock otherwise
7131 we can end up in a deadlock if another thread is waiting for the lock
7132 and media unprepare is also waiting for that thread to end.
7133 https://bugzilla.gnome.org/show_bug.cgi?id=737690
7135 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
7137 * gst/rtsp-server/rtsp-media.c:
7138 rtsp-media: Set state to UNPREPARING in all cases
7140 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
7142 * gst/rtsp-server/rtsp-media.c:
7143 media: set state to unpreparing when unprepare is initiated
7144 https://bugzilla.gnome.org/show_bug.cgi?id=737675
7146 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
7148 * gst/rtsp-server/rtsp-client.c:
7149 rtsp-client: Remove backlog limit while processings requests
7150 If the backlog limit is kept two cases of deadlocks may be
7151 encountered when streaming over TCP. Without the backlog
7152 limit this deadlocks can not happen, at the expence of
7154 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
7156 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
7158 * gst/rtsp-server/rtsp-client.c:
7159 rtsp-client: do not free main context before rtsp watch
7160 https://bugzilla.gnome.org/show_bug.cgi?id=737110
7162 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
7164 * tests/check/gst/rtspserver.c:
7165 tests: Extend unit test timeout to accomodate for valgrind
7166 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7168 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
7170 * gst/rtsp-server/rtsp-client.c:
7171 * gst/rtsp-server/rtsp-session.c:
7172 * gst/rtsp-server/rtsp-stream-transport.c:
7173 rtsp-*: Treat sending packets to clients as keepalive
7174 As long as gst-rtsp-server can successfully send RTP/RTCP data to
7175 clients then the client must be reading. This change makes the server
7176 timeout the connection if the client stops reading.
7177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7179 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
7181 * gst/rtsp-server/rtsp-client.c:
7182 rtsp-client: Allow backlog to grow while expiring session
7183 Allow the send backlog in the RTSP watch to grow to unlimited size while
7184 attempting to bring the media pipeline to NULL due to a session
7185 expiring. Without this change the appsink element cannot change state
7186 because it is blocked while rendering data in the new_sample callback.
7187 This callback will block until it has successfully put the data into the
7188 send backlog. There is a chance that the send backlog is full at this
7189 point which means that the callback may block for a long time, possibly
7190 forever. Therefore the media pipeline may also be prevented from
7191 changing state for a long time.
7192 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7194 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
7196 * gst/rtsp-server/rtsp-client.c:
7197 rtsp-client: Make old compilers happy
7198 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
7199 Just in case that guint8 doesn't fit in a pointer. Just in case ...
7201 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
7203 * gst/rtsp-server/rtsp-client.c:
7204 client: raise the backlog limits before pausing
7205 We need to raise the backlog limits before pausing the pipeline or else
7206 the appsink might be blocking in the render method in wait_backlog() and
7207 we would deadlock waiting for paused.
7208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
7210 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
7212 * gst/rtsp-server/rtsp-client.c:
7213 client: make define for the WATCH_BACKLOG
7214 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
7216 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
7218 * gst/rtsp-server/rtsp-client.c:
7219 client: simplify session transport handling
7220 link/unlink of the transport in a session was done to keep track of all
7221 TCP transports and to send RTP/RTCP data to the streams. We can simplify
7222 that by putting all the TCP transports in a hashtable indexed with the
7224 We also don't need to link/unlink the transports when we pause/resume
7225 the streams. The same effect is already achieved when we pause/play the
7226 media. Indeed, when we pause the media, the transport is removed from
7227 the media and the callbacks will not be called anymore.
7228 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
7230 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
7232 * gst/rtsp-server/rtsp-stream-transport.c:
7233 * gst/rtsp-server/rtsp-stream-transport.h:
7234 stream-transport: make method to handle received data
7235 Make a method to handle the data received on a channel. It sends the
7236 data to the stream of the transport on the RTP or RTCP pads based on
7239 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
7241 * examples/test-mp4.c:
7242 test: add example of dumping RTCP reports
7244 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
7246 * gst/rtsp-server/rtsp-media.c:
7247 * gst/rtsp-server/rtsp-stream.c:
7248 * gst/rtsp-server/rtsp-stream.h:
7249 rtsp-media: Make sure that sequence numbers are monotonic after pause
7250 The sequence number is not monotonic for RTP packets after pause. The
7251 reason is basepayloader generates a randon sequence number when the
7252 pipeline goes from ready to pause. With this fix generation of sequence
7253 number will be monotonic when going from pause to play request.
7254 https://bugzilla.gnome.org/show_bug.cgi?id=736017
7256 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
7258 * gst/rtsp-server/rtsp-client.c:
7259 rtsp-client: Protect saved clients watch with a mutex
7260 Fixes a crash when close() is called while merging clients
7261 in handle_tunnel(). In that case close() would destroy the
7262 watch while it is still being used in handle_tunnel().
7263 https://bugzilla.gnome.org/show_bug.cgi?id=735570
7265 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
7267 * gst/rtsp-server/rtsp-stream.c:
7268 rtsp-stream: Remove the multicast group udp sources when removing from the bin
7270 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
7272 * gst/rtsp-server/rtsp-media.c:
7273 * gst/rtsp-server/rtsp-stream.c:
7274 * gst/rtsp-server/rtsp-stream.h:
7275 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
7276 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
7277 seeking and will always continue counting the time. This leads to
7278 the NPT after a backwards seek to be something completely different
7279 to the actual seek position.
7280 https://bugzilla.gnome.org/show_bug.cgi?id=732644
7282 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
7284 * examples/test-appsrc.c:
7285 examples: fix another reference leak
7286 gst_rtsp_media_get_element() returns a new ref.
7288 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
7290 * examples/test-appsrc.c:
7291 examples: unref element after usage
7292 gst_bin_get_by_name_recurse_up() returns an element
7293 reference that must be unreffed after usage.
7294 https://bugzilla.gnome.org/show_bug.cgi?id=734546
7296 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
7298 * gst/rtsp-server/rtsp-media.c:
7299 signals: Fix copy-pasto in target-state signal offset
7301 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
7305 Makefile: Add usage of build-checks step
7306 Allows building checks without running them
7308 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
7310 * gst/rtsp-server/rtsp-stream.c:
7311 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
7312 When a UDP multicast transport is used it is expected that the server listens
7313 for RTP and RTCP packets on the multicast group with the corresponding port.
7314 Without this we will never get RTCP packets from clients in multicast mode.
7315 https://bugzilla.gnome.org/show_bug.cgi?id=732238
7317 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
7322 === release 1.4.0 ===
7324 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7330 * gst-rtsp-server.doap:
7333 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
7335 * gst/rtsp-server/rtsp-media.h:
7336 media: correct misspelled words in description
7337 https://bugzilla.gnome.org/show_bug.cgi?id=733244
7339 === release 1.3.91 ===
7341 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
7347 * gst-rtsp-server.doap:
7350 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
7352 * docs/libs/gst-rtsp-server-sections.txt:
7355 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
7357 * gst/rtsp-server/rtsp-server.c:
7358 server: implement client REMOVE filter
7360 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
7362 * gst/rtsp-server/rtsp-client.c:
7363 * gst/rtsp-server/rtsp-client.h:
7364 client: expose _close() method
7365 Expose a previously internal close method to close the client
7368 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
7370 * gst/rtsp-server/rtsp-session-pool.c:
7371 session-pool: signal session-removed outside of the lock
7372 Release the lock before emiting the session-removed signal.
7374 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
7376 * gst/rtsp-server/rtsp-client.c:
7377 * gst/rtsp-server/rtsp-server.c:
7378 * gst/rtsp-server/rtsp-session-pool.c:
7379 * gst/rtsp-server/rtsp-session.c:
7380 * gst/rtsp-server/rtsp-stream.c:
7381 filter: Release lock in filter functions
7382 Release the object lock before calling the filter functions. We need to
7383 keep a cookie to detect when the list changed during the filter
7384 callback. We also keep a hashtable to make sure we only call the filter
7385 function once for each object in case of concurrent modification.
7386 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7388 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7390 * gst/rtsp-server/rtsp-client.c:
7391 client: check if watch is set in handle_teardown()
7392 The unit tests run without a watch
7394 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7396 * tests/check/gst/client.c:
7397 client tests: send teardown to cleanup session
7399 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7401 * tests/check/gst/rtspserver.c:
7402 server tests: send teardown to cleanup session
7404 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7406 * gst/rtsp-server/rtsp-client.c:
7407 client: keep ref to client for the session removed handler
7408 This extra ref will be dropped when all client sessions have been
7409 removed. A session is removed when a client sends teardown, closes its
7410 endpoint of the TCP connection or the sessions expires.
7411 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7413 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7415 * gst/rtsp-server/rtsp-client.c:
7416 * gst/rtsp-server/rtsp-session.c:
7417 * tests/check/gst/client.c:
7418 client: manage media in session as a last step
7419 Once we manage a media in a session, we can't unmanage it anymore
7420 without destroying it. Therefore, first check everything before we
7421 manage the media, otherwise if something is wrong we have no way to
7423 If we created a new session and something went wrong, remove the session
7424 again. Fixes a leak in the unit test.
7426 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7428 * examples/test-mp4.c:
7429 * examples/test-ogg.c:
7430 examples: print 'stream ready at url' for mp4 and ogg example
7432 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7434 * gst/rtsp-server/rtsp-client.c:
7435 * gst/rtsp-server/rtsp-sdp.c:
7436 rtsp: fix for MIKEY api change
7438 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7440 * gst/rtsp-server/rtsp-client.c:
7441 client: free watch context only once
7442 The watch context is freed when the source is destroyed. Avoids
7443 a CRITICAL when we try to unref the context twice.
7445 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7447 * gst/rtsp-server/rtsp-client.c:
7450 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7452 * gst/rtsp-server/rtsp-client.c:
7453 client: protect sessions with lock
7454 Protect the list of sessions with the lock.
7455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7457 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7459 * gst/rtsp-server/rtsp-client.c:
7460 Client: keep a ref to the session
7461 Don't just keep a weak ref to the session objects but use a hard ref. We
7462 will be notified when a session is removed from the pool (expired) with
7463 the new session-removed signal.
7464 Don't automatically close the RTSP connection when all the sessions of
7465 a client are removed, a client can continue to operate and it can create
7466 a new session if it wants. If you want to remove the client from the
7467 server, you have to use gst_rtsp_server_client_filter() now.
7468 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7469 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7471 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7473 * gst/rtsp-server/rtsp-session-pool.c:
7474 * gst/rtsp-server/rtsp-session-pool.h:
7475 session-pool: add session-removed signal
7476 Add a signal to be notified when a session is removed from the pool.
7478 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7480 * gst/rtsp-server/Makefile.am:
7481 * gst/rtsp-server/rtsp-server.h:
7482 Make rtsp-server.h a single-include header, use it for G-I
7483 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7485 === release 1.3.90 ===
7487 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7493 * gst-rtsp-server.doap:
7496 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7498 * gst/rtsp-server/rtsp-stream.c:
7499 stream: crypto can be NULL
7501 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7503 * gst/rtsp-server/rtsp-client.c:
7504 * gst/rtsp-server/rtsp-media.c:
7505 * gst/rtsp-server/rtsp-mount-points.c:
7506 introspection: add missing allow-none annotations
7507 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7509 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7511 * gst/rtsp-server/rtsp-address-pool.c:
7512 * gst/rtsp-server/rtsp-media.c:
7513 * gst/rtsp-server/rtsp-session-media.c:
7514 * gst/rtsp-server/rtsp-session-pool.c:
7515 * gst/rtsp-server/rtsp-stream-transport.c:
7516 * gst/rtsp-server/rtsp-stream.c:
7517 * gst/rtsp-server/rtsp-token.c:
7518 introspection: add (nullable) annotations to return values
7519 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7521 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7523 * gst/rtsp-server/rtsp-client.c:
7524 * gst/rtsp-server/rtsp-stream.c:
7525 gi: improve annotations
7526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7528 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7530 * gst/rtsp-server/rtsp-client.c:
7531 * gst/rtsp-server/rtsp-media-factory.c:
7532 * gst/rtsp-server/rtsp-media.c:
7533 * gst/rtsp-server/rtsp-server.c:
7534 signals: use generic marshal function
7535 Use the generic C marshal function.
7536 Use more explicit type instead of G_TYPE_POINTER
7538 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7540 * gst/rtsp-server/rtsp-context.h:
7541 context: add type macro
7543 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7545 * gst/rtsp-server/rtsp-client.c:
7546 * gst/rtsp-server/rtsp-sdp.c:
7547 * gst/rtsp-server/rtsp-sdp.h:
7548 sdp: hide key length defines
7549 They don't have a namespace.
7551 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7556 === release 1.3.3 ===
7558 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7564 * gst-rtsp-server.doap:
7567 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7569 * gst/rtsp-server/rtsp-client.c:
7570 * gst/rtsp-server/rtsp-sdp.c:
7571 * gst/rtsp-server/rtsp-sdp.h:
7572 mikey: add different key length parameters
7573 Add encryption and authentication key length parameters to MIKEY. For
7574 the encoders, the key lengths are obtained from the cipher and auth
7575 algorithms set in the caps. For the decoders, they are obtained while
7576 parsing the key management from the client.
7577 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7579 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7581 * tests/check/gst/stream.c:
7582 stream tests: Make sure we get right multicast address from stream
7583 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7585 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7587 * gst/rtsp-server/rtsp-client.c:
7588 client: ref the context until rtsp watch is alive
7589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7591 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7593 * gst/rtsp-server/rtsp-client.c:
7594 client: Destroy the rtsp watch after connection close
7596 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7598 * gst/rtsp-server/rtsp-media.c:
7599 media: fix confusing comment
7601 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7603 * gst/rtsp-server/rtsp-session.c:
7604 rtsp-session: Timeout in header.
7605 Adding the possbilty to always have timout in header.
7606 This is configurabe with setting "timeout-always-visible".
7607 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7609 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7614 === release 1.3.2 ===
7616 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7623 * gst-rtsp-server.doap:
7626 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7629 Automatic update of common submodule
7630 From 211fa5f to 1f5d3c3
7632 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7634 * gst/rtsp-server/rtsp-client.c:
7635 client: store TCP ports in transport
7636 Store the TCP ports in the transport when we are doing RTSP over TCP.
7637 This way, we can easily get to the ports from the transport.
7638 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7640 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7642 * gst/rtsp-server/rtsp-stream.c:
7643 stream: add signals for new RTP/RTCP encoders
7644 New signals to allow the user to configure the dynamically created
7646 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7648 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7650 * gst/rtsp-server/rtsp-media.c:
7651 * gst/rtsp-server/rtsp-media.h:
7652 media: Make suspend()/unsuspend() virtual
7653 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7655 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7657 * gst/rtsp-server/rtsp-client.c:
7658 client: fix send-message signal marshaller
7659 Use generic marshalling for the send-message signal. It has
7660 two POINTER arguments, not just one.
7661 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7663 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7665 * tests/check/gst/media.c:
7666 tests: add and remove pads only once
7667 In this test we simulate a dynamic pad by watching the caps event.
7668 Because of renegotiation in the base payloader now, this caps is sent
7669 multiple times but we can only deal with 1 invocation, use a variable to
7670 only 'add and remove' the pad once.
7672 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7674 * tests/check/gst/rtspserver.c:
7675 tests: add unit test for correct handling of Require headers
7676 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7678 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7680 * gst/rtsp-server/rtsp-client.c:
7681 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7682 Servers must handle Require headers and must report a failure
7683 if they don't handle any of the Required options, see RFC 2326,
7684 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7685 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7687 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7692 === release 1.3.1 ===
7694 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7700 * gst-rtsp-server.doap:
7703 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7706 Automatic update of common submodule
7707 From bcb1518 to 211fa5f
7709 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7714 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7716 * tests/check/gst/sessionmedia.c:
7717 tests: fix memory leak in sessionmedia unit test
7719 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7721 * gst/rtsp-server/rtsp-client.c:
7722 client: emit a signal before sending a message
7723 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7725 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7727 * gst/rtsp-server/rtsp-client.c:
7728 client: pass context to send_message
7729 Pass the current context to send_message, we will need it later.
7731 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7733 * gst/rtsp-server/rtsp-client.c:
7734 client: fix typo in comment
7736 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7738 * gst/rtsp-server/rtsp-media.c:
7739 media: Do not stop thread twice if default_prepare() fails
7741 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7743 * gst/rtsp-server/rtsp-client.c:
7744 client: set the watch to flushing before going to NULL
7745 First set the watch to flushing so that we unblock any current and
7746 future attempt to send data on the watch, Then set the pipeline to
7748 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7750 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7752 * gst/rtsp-server/rtsp-session-pool.c:
7753 * tests/check/gst/sessionpool.c:
7754 rtsp-session-pool: Fixes annotation
7755 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7756 in the sessionpool test.
7757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7759 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7761 * gst/rtsp-server/rtsp-media.c:
7762 * gst/rtsp-server/rtsp-media.h:
7763 media: make media_prepare virtual
7764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7766 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7768 * gst/rtsp-server/rtsp-media.c:
7769 * tests/check/gst/media.c:
7770 media: stop the thread in more error cases
7772 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7774 * gst/rtsp-server/rtsp-media.c:
7775 * tests/check/gst/media.c:
7776 media: allow NULL as the thread
7777 Use the default context whan passing a NULL thread.
7779 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7781 * gst/rtsp-server/rtsp-client.c:
7782 rtsp-client: indent cleanup
7783 Coverity was moaning about unreachable code, and I think it was just
7784 confused by { being before the label. We'll see if it pops up again.
7787 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7789 * gst/rtsp-server/rtsp-client.c:
7790 * gst/rtsp-server/rtsp-media.c:
7791 client: Add drop-backlog property
7792 When we have too many messages queued for a client (currently hardcoded
7793 to 100) we overflow and drop the messages. Add a drop-backlog property
7794 to control this behaviour. Setting this property to FALSE will retry
7795 to send the messages to the client by waiting for more room in the
7797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7799 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7801 * gst/rtsp-server/rtsp-client.c:
7802 client: support for POST before GET when setting up a tunnel
7804 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7806 * gst/rtsp-server/rtsp-client.c:
7807 client: remove watch of the second client after http tunnel setup
7808 The second client will be freed after the HTTP tunnel has been set up.
7809 Make sure it's RTSP watch is never dispatched again.
7810 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7812 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7814 * gst/rtsp-server/rtsp-media.c:
7815 * tests/check/gst/media.c:
7816 media: Make media_prepare() fail if port allocation fails
7817 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7819 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7821 * tests/check/gst/media.c:
7822 media test: cleanup the thread pool in tests
7824 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7826 * gst/rtsp-server/rtsp-media.c:
7827 * tests/check/gst/media.c:
7828 rtsp-media: Unblock blocked streams in unprepare
7829 The streams will be blocked when a live media is prepared.
7830 The streams should be unblocked in gst_rtsp_media_unprepare.
7831 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7833 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7835 * gst/rtsp-server/rtsp-media.c:
7836 media: release the state lock when going to NULL
7837 Set our state to UNPREPARING and release the state-lock before
7838 setting the pipeline to the NULL state. This way, any pad-added
7839 callback will be able to take the state-lock and check that we are now
7840 unpreparing instead of deadlocking.
7841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7843 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7845 * gst/rtsp-server/rtsp-media.c:
7846 media: protect status with lock
7847 Make sure we only update the status with the lock.
7849 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7851 * gst/rtsp-server/rtsp-client.c:
7852 * gst/rtsp-server/rtsp-sdp.c:
7853 rtsp: update for MIKEY API changes
7855 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7857 * gst/rtsp-server/rtsp-client.c:
7858 client: parse the mikey response from the client
7859 Parse the mikey response from the client and update the policy for
7862 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7864 * gst/rtsp-server/rtsp-stream.c:
7865 * gst/rtsp-server/rtsp-stream.h:
7866 stream: add method to set crypto info
7867 Make a method to configure the crypto information of a stream.
7868 Set udpsrc in READY instead of PAUSED so that we can configure caps
7871 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7873 * gst/rtsp-server/rtsp-client.c:
7874 client: cleanup error paths
7876 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7878 * gst/rtsp-server/rtsp-media.c:
7881 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7883 * examples/test-video.c:
7884 test: enable SRTP only on RTSPS
7885 We only want to enable SRTP when doing rtsp over TLS so that we can
7886 exchange the keys in a secure way.
7888 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7890 * examples/test-video.c:
7891 test: print an error on failure
7893 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7896 * examples/test-video.c:
7897 * gst/rtsp-server/rtsp-sdp.c:
7898 * gst/rtsp-server/rtsp-stream.c:
7899 * tests/check/Makefile.am:
7900 stream: add SRTP support
7901 Install srtp encoder and decoder elements in rtpbin
7904 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7906 * tests/check/Makefile.am:
7907 * tests/check/gst/sessionpool.c:
7908 tests: Add unit tests for sessionpool
7909 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7911 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7913 * tests/check/gst/threadpool.c:
7914 tests: Improve code coverage of rtsp-threadpool tests
7915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7917 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7919 * tests/check/gst/sessionmedia.c:
7920 tests: Improve code coverage for rtsp-session-media
7921 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7923 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7925 gobject-introspection: Add annotations to support language bindings
7926 In addition a few cosmetic changes:
7927 * Adjust the order of arguments
7928 * Fix typo: occured -> occurred
7929 * Fix indentation after Return:-clauses
7930 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7932 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7934 * gst/rtsp-server/rtsp-stream.c:
7935 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7936 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7938 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7940 * gst/rtsp-server/rtsp-stream.c:
7941 stream: take caps after the session manager
7942 Take the caps for the SDP after they leave the rtpbin so that we can
7943 also get the properties added by rtpbin elements.
7945 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7947 * gst/rtsp-server/rtsp-stream.c:
7948 stream: release lock while pushing out packets
7949 Keep a cache of the transports and use this to iterate the transport
7950 while pushing packets. This allows us to release the lock early.
7951 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7953 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7955 * gst/rtsp-server/rtsp-client.c:
7956 * gst/rtsp-server/rtsp-client.h:
7957 rtsp-client: vmethod for modifying tunnel GET response
7958 Add a vmethod tunnel_http_response where the response to the HTTP GET
7959 for tunneled connections can be modified.
7960 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7962 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7964 * gst/rtsp-server/rtsp-sdp.c:
7965 sdp: make 1 media line per profile
7966 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7967 line in the SDP for each profile. The client is then supposed to pick
7968 one of the profiles in the SETUP request. Because the m= lines have the
7969 same pt, the client also knows that only 1 option is possible.
7971 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7973 * gst/rtsp-server/rtsp-media-factory.c:
7974 * gst/rtsp-server/rtsp-media-factory.h:
7975 * gst/rtsp-server/rtsp-media.c:
7976 factory: add profile property and pass to media and streams
7978 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7980 * examples/test-multicast.c:
7981 * gst/rtsp-server/rtsp-sdp.c:
7982 sdp: pass multicast connection for multicast-only stream
7983 Pass the multicast address of the stream in the connection info in the
7984 SDP so that clients try a multicast connection first.
7985 Only allow multicast connections in the test-multicast example. Also
7986 increase the TTL a little.
7988 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7991 .gitignore: Ignore gcov intermediate files
7992 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7994 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7996 * gst/rtsp-server/rtsp-stream.c:
7997 stream: release some locks in error cases
7999 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8001 docs: Enable and fix gtk-doc warnings
8002 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
8003 * addresspool/mediafactory: Add missing annotation colon
8004 * stream: Annotate return value
8005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
8007 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
8010 Automatic update of common submodule
8011 From fe1672e to bcb1518
8013 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
8016 Automatic update of common submodule
8017 From 1a07da9 to fe1672e
8019 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
8021 * examples/Makefile.am:
8022 examples: use LDADD for libs instead of LDFLAGS
8024 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
8027 configure: make sure releases are in .doap file
8029 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
8031 * examples/test-cgroups.c:
8032 examples: test-cgroups: don't put code with side effects into g_assert()
8033 The g_assert() might get compiled out with the right
8034 compiler/preprocessor flags.
8036 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
8038 * examples/.gitignore:
8039 examples: add cgroup test binary to .gitignore
8041 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
8043 * examples/test-cgroups.c:
8044 examples: fix cgroup test build
8045 Fixes build failure caused by compiler warning:
8046 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
8048 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
8051 .gitignore: ignore temp files created in the course of 'make check'
8053 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
8055 * gst/rtsp-server/rtsp-media.c:
8056 rtsp-media: don't loose frames handling new PLAY request
8057 If client supplied a range check if the range specifies the start point.
8058 If not, then do an accurate seek to the current position. If a start
8059 point was specified do do a key unit seek to make sure the streaming
8060 starts with decodeable frames.
8061 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
8063 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
8065 * gst/rtsp-server/rtsp-media.c:
8066 Revert "media: only flush when setting a new start position"
8067 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
8068 We need to do the flush in all cases, demuxer block currently for
8071 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
8073 * gst/rtsp-server/rtsp-media.c:
8074 media: only flush when setting a new start position
8075 Only flush the pipeline when we change the start position with
8077 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
8079 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
8081 * gst/rtsp-server/rtsp-stream.c:
8082 stream: set ttl-mc before adding the socket
8083 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
8084 never be set on socket.
8085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
8087 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8089 * gst/rtsp-server/rtsp-media.c:
8090 media: stop thread if media is already prepared
8091 in gst_rtsp_media_prepare() the thread is not used if media is already
8092 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
8094 https://bugzilla.gnome.org/show_bug.cgi?id=724182
8096 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
8099 build: Ship gst-rtsp-server.doap file
8101 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
8103 * tests/check/gst/rtspserver.c:
8104 tests: Fix another compiler warning with gcc
8106 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
8108 * gst/rtsp-server/rtsp-client.c:
8109 * gst/rtsp-server/rtsp-mount-points.c:
8110 * gst/rtsp-server/rtsp-stream.c:
8111 * tests/check/gst/client.c:
8112 rtsp-server: Fix lots of compiler warnings with clang
8114 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
8117 * gst-rtsp-server.doap:
8118 * tests/Makefile.am:
8119 configure: Synchronise with the configure scripts of the other modules
8121 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
8124 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
8126 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
8128 * gst/rtsp-server/rtsp-media.c:
8129 * gst/rtsp-server/rtsp-stream.c:
8130 Revert "rtsp-server: support build against last stable release"
8131 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
8132 Let us require 1.2.3 now, which is going to be released in a few
8135 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
8137 * gst/rtsp-server/rtsp-session-media.c:
8138 * gst/rtsp-server/rtsp-stream-transport.c:
8139 session: improve RTP-Info
8140 Ignore streams that can't generate RTP-Info instead of failing.
8141 Don't return the empty string when all streams are unconfigured but
8142 return NULL so that we don't generate and empty RTP-Info header.
8143 Improve docs a little.
8145 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
8147 * gst/rtsp-server/rtsp-session-media.c:
8148 Don't free rtpinfo GString when it is NULL
8149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8151 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
8153 * gst/rtsp-server/rtsp-media.c:
8154 media: only set keyframe flag when modifying start
8155 Only set the keyframe flag when we modify the start position. The
8156 keyframe flag should probably be ignored when no change is requested but
8157 until we can claim this is all documented properly and all demuxer
8158 implement this, avoid setting the flag.
8159 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
8161 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
8163 * gst/rtsp-server/rtsp-thread-pool.c:
8164 thread-pool: Unref source after mainloop has quit to avoid races in GLib
8165 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
8167 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
8169 * gst/rtsp-server/rtsp-stream.c:
8170 stream: handle NULL seqnum and rtptime arguments
8172 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
8174 * gst/rtsp-server/rtsp-thread-pool.c:
8175 * tests/check/gst/threadpool.c:
8176 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
8177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
8179 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
8181 * gst/rtsp-server/rtsp-stream.c:
8182 stream: add fallback for missing stats property
8183 Use a fallback when the payloader does not have a stats property
8184 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8186 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
8189 Automatic update of common submodule
8190 From f7bc1c3 to 1a07da9
8192 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
8194 * gst/rtsp-server/rtsp-stream.c:
8195 stream: don't leak stats structure
8196 Don't leak the stats structure and deal with NULL stats.
8198 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
8200 * gst/rtsp-server/rtsp-stream.c:
8201 stream: Get rtpinfo properties atomically from payloader
8202 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
8204 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
8206 * gst/rtsp-server/rtsp-media.c:
8207 media: refactor state change functions and signals
8208 Make functions to set the target state and the pipeline state and emit
8209 the signals from those functions.
8211 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
8213 * gst/rtsp-server/rtsp-media.c:
8214 * gst/rtsp-server/rtsp-media.h:
8215 media: add signal to notify of pending state changes
8217 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
8219 * gst/rtsp-server/rtsp-media.c:
8220 * gst/rtsp-server/rtsp-stream.c:
8221 rtsp-server: support build against last stable release
8222 Until 1.2.3 is out with the new get_type function and we
8225 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
8227 * gst/rtsp-server/rtsp-stream.c:
8228 stream: fix compilation
8230 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
8232 * gst/rtsp-server/rtsp-media.c:
8233 * gst/rtsp-server/rtsp-media.h:
8234 * gst/rtsp-server/rtsp-stream.c:
8235 * gst/rtsp-server/rtsp-stream.h:
8236 stream: add property to configure profiles
8238 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
8240 * gst/rtsp-server/rtsp-client.c:
8241 client: let stream check supported transport
8242 Delegate the check if a transport is allowed to the stream.
8243 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
8245 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
8247 * gst/rtsp-server/rtsp-stream.c:
8248 * gst/rtsp-server/rtsp-stream.h:
8249 stream: add method to check supported transport
8250 Add a method to check if a transport is supported
8252 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
8255 configure.ac: Only check for gstreamer-check, not check
8256 We include check in gstreamer-check since quite some time now.
8258 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
8260 * gst/rtsp-server/rtsp-session-media.c:
8261 * gst/rtsp-server/rtsp-stream-transport.c:
8262 * gst/rtsp-server/rtsp-stream.c:
8263 * gst/rtsp-server/rtsp-stream.h:
8264 stream: return clock-rate from get_rtpinfo
8265 And use it to correct the rtptime to the requested start-time.
8266 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
8268 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
8270 * gst/rtsp-server/rtsp-session-media.c:
8271 * gst/rtsp-server/rtsp-stream-transport.c:
8272 * gst/rtsp-server/rtsp-stream-transport.h:
8273 session-media: calculate start-time
8275 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
8277 * gst/rtsp-server/rtsp-stream-transport.c:
8278 * gst/rtsp-server/rtsp-stream.c:
8279 * gst/rtsp-server/rtsp-stream.h:
8280 stream: also return the running-time
8281 Return the running-time in the rtpinfo as well.
8283 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
8285 * gst/rtsp-server/rtsp-client.c:
8286 * gst/rtsp-server/rtsp-session-media.c:
8287 * gst/rtsp-server/rtsp-session-media.h:
8288 * gst/rtsp-server/rtsp-stream-transport.c:
8289 * gst/rtsp-server/rtsp-stream-transport.h:
8290 session-media: let the session-media make the RTPInfo
8291 Add method to create the RTPInfo for a stream-transport.
8292 Add method to create the RTPInfo for all stream-transports in a
8294 Use the session-media RTPInfo code in client. This allows us to refactor
8295 another method to link the TCP callbacks.
8297 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8299 mount-points: sort sequence before g_sequence_lookup
8300 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
8301 sort sequence if dirty, otherwise lookup will fail.
8302 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
8304 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
8307 configure: rename package from gst-rtsp to gst-rtsp-server
8308 To match git module name and avoid confusion with the
8309 rtsp lib in gst-plugins-base and rtsp plugin in -good.
8311 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
8314 configure: bump core/base/good requirement to 1.2.0
8315 Bump to released stable version and make implicit
8316 requirements explicit.
8318 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
8323 Fix broken gettext setup which is not used anyway
8325 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
8328 Automatic update of common submodule
8329 From dbedaa0 to d48bed3
8331 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
8333 * gst/rtsp-server/rtsp-client.c:
8334 * gst/rtsp-server/rtsp-media.c:
8335 * gst/rtsp-server/rtsp-media.h:
8336 media: add setup_sdp vmethod
8337 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
8338 gst_rtsp_media_setup_sdp.
8339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
8341 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
8343 * gst/rtsp-server/rtsp-stream.c:
8344 rtsp-stream: Check return value of sscanf
8345 streamid is only valid if sscanf matched something.
8347 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
8349 * gst/rtsp-server/rtsp-client.c:
8350 rtsp-client: Fix iteration
8351 Wouldn't even enter the code block otherwise (i++ was used as the check
8352 and not the postfix).
8354 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
8356 * gst/rtsp-server/rtsp-client.c:
8357 * gst/rtsp-server/rtsp-client.h:
8358 client: add vmethod to configure media and streams
8359 Implement a vmethod that can be used to configure the media and the
8360 streams based on the current context. Handle the blocksize handling in
8361 the default handler.
8362 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
8364 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8367 Make git ignore more unit test binaries
8369 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8371 * gst/rtsp-server/rtsp-address-pool.h:
8372 * gst/rtsp-server/rtsp-auth.h:
8373 * gst/rtsp-server/rtsp-client.h:
8374 * gst/rtsp-server/rtsp-context.h:
8375 * gst/rtsp-server/rtsp-media-factory-uri.h:
8376 * gst/rtsp-server/rtsp-media-factory.h:
8377 * gst/rtsp-server/rtsp-media.h:
8378 * gst/rtsp-server/rtsp-mount-points.h:
8379 * gst/rtsp-server/rtsp-server.h:
8380 * gst/rtsp-server/rtsp-session-media.h:
8381 * gst/rtsp-server/rtsp-session-pool.h:
8382 * gst/rtsp-server/rtsp-session.h:
8383 * gst/rtsp-server/rtsp-stream-transport.h:
8384 * gst/rtsp-server/rtsp-stream.h:
8385 * gst/rtsp-server/rtsp-thread-pool.h:
8386 * gst/rtsp-server/rtsp-token.h:
8387 rtsp-server: add padding to many public structures
8388 Not mini objects though, since they are not subclassable
8389 anyway, nor kept on the stack or inlined in a structure.
8391 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8393 media: add new create_rtpbin vmethod
8394 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8395 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8397 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8399 * tests/check/gst/media.c:
8400 tests: fix memory leak, free test's thread pool
8401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8403 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8405 * gst/rtsp-server/rtsp-stream-transport.c:
8406 stream-transport: free url in finalize
8408 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8410 * gst/rtsp-server/rtsp-media.c:
8411 media: also do state change in suspended state
8413 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8415 * gst/rtsp-server/rtsp-client.c:
8416 * gst/rtsp-server/rtsp-media.c:
8417 media: also handle prepare and range in suspended state
8418 When we are suspended, we are already prepared.
8419 We can get the range in the suspended state.
8421 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8423 * tests/check/Makefile.am:
8424 * tests/check/gst/sessionmedia.c:
8425 check: add test for uri in setup
8426 Added unit tests for the new functionality in GstRTSPStreamTransport.
8427 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8429 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8431 * gst/rtsp-server/rtsp-client.c:
8432 client: store setup uri and use in PLAY response
8433 Store the uri used when doing the setup and use that in the PLAY
8435 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8437 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8439 * gst/rtsp-server/rtsp-stream-transport.c:
8440 * gst/rtsp-server/rtsp-stream-transport.h:
8441 stream-transport: add method to get/set url
8443 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8445 * gst/rtsp-server/rtsp-client.c:
8446 client: suspend after SDP and unsuspend before PLAYING
8447 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8450 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8452 * gst/rtsp-server/rtsp-media-factory.c:
8453 * gst/rtsp-server/rtsp-media-factory.h:
8454 * gst/rtsp-server/rtsp-media.c:
8455 * gst/rtsp-server/rtsp-media.h:
8456 * gst/rtsp-server/rtsp-session-media.c:
8457 * gst/rtsp-server/rtsp-session.c:
8458 * tests/check/gst/media.c:
8459 * tests/check/gst/mediafactory.c:
8460 media: add suspend modes
8461 Add support for different suspend modes. The stream is suspended right after
8462 producing the SDP and after PAUSE. Different suspend modes are available that
8463 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8464 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8465 state and RESET will bring the pipeline to the NULL state.
8466 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8467 this means that the pipeline needs to be prerolled again.
8468 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8469 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8471 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8473 * gst/rtsp-server/rtsp-media.c:
8474 media: start live streams in blocked state
8475 Start live streams in the blocked state and make them preroll using the
8476 messages. This ensure that no data is played by the sink until we explicitly
8477 unblock the stream right before going to PLAYING.
8478 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8480 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8482 * gst/rtsp-server/rtsp-media.c:
8483 media: refactor starting and waiting for preroll
8484 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8485 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8487 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8489 * gst/rtsp-server/rtsp-stream.c:
8490 * gst/rtsp-server/rtsp-stream.h:
8491 stream: add API to block streams
8492 Add an API to block on the streams and make it post a message.
8493 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8494 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8496 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8498 * docs/libs/Makefile.am:
8499 docs: Specify the override file
8500 Even if it's empty (for now) it avoids make distcheck complaining
8502 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8504 * gst/rtsp-server/rtsp-media.c:
8505 media: move default implementations to where they are used
8507 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8509 * gst/rtsp-server/rtsp-media.c:
8510 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8511 We need to take the state_lock when calling this method.
8513 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8515 * gst/rtsp-server/rtsp-media.c:
8516 media: handle add-added on non-bins too
8517 Handle dynamic payloaders that are not bins, as used in the unit-test.
8519 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8521 * gst/rtsp-server/rtsp-media-factory.c:
8522 * gst/rtsp-server/rtsp-media-factory.h:
8523 * gst/rtsp-server/rtsp-media.c:
8524 rtsp-media/-factory: Fix request pad name comments
8525 These must be escaped for gtk-doc to parse the comments without warnings.
8527 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8529 rtsp-media: remove transports if media is in error status
8530 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8531 trying to change to GST_STATE_NULL and media is in error status, we
8532 remove all transports.
8533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8535 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8537 * gst/rtsp-server/rtsp-media.c:
8538 rtsp-media: use element metadata to find payloader
8539 Use the element metadata to find the payloader instead of checking
8541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8543 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8545 rtsp-stream: add getter for payload type
8546 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8547 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8548 element and create the stream with this one instead of the dynpay%d
8550 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8552 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8554 * gst/rtsp-server/rtsp-client.c:
8555 * gst/rtsp-server/rtsp-context.h:
8556 * gst/rtsp-server/rtsp-media.c:
8557 * gst/rtsp-server/rtsp-mount-points.c:
8558 * gst/rtsp-server/rtsp-server.c:
8559 * gst/rtsp-server/rtsp-token.c:
8560 rtsp-*: Refer to NULL as a constant in comments
8562 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8564 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8566 rtsp-*: Fix type name typos in comments
8567 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8568 * rtsp-auth: Refer to part of constant name as text
8569 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8570 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8571 * rtsp-stream: Fix typo when refering to GstBin
8572 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8574 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8577 * docs/libs/gst-rtsp-server-docs.sgml:
8578 * docs/libs/gst-rtsp-server-sections.txt:
8579 docs: Improve documentation
8580 * Include annotation-glossary to quiet gtk-doc
8581 * Rename remaining ClientState -> Context
8582 * Rename object hierarchy file
8583 * Remove stale chapter references
8584 * Add missing function and object references
8585 * Include missing GstRTSPAddressPoolResult
8586 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8588 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8590 * gst/rtsp-server/rtsp-client.c:
8591 * gst/rtsp-server/rtsp-server.c:
8592 * gst/rtsp-server/rtsp-session-pool.c:
8593 * gst/rtsp-server/rtsp-session.c:
8594 * gst/rtsp-server/rtsp-stream.c:
8595 rtsp-server: sprinkle some allow-none annotations for g-i
8597 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8599 * gst/rtsp-server/rtsp-stream.c:
8600 * gst/rtsp-server/rtsp-stream.h:
8601 stream: add method to filter transports
8602 Add a method to safely iterate and collect the stream transports
8603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8605 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8607 * gst/rtsp-server/rtsp-client.c:
8608 * gst/rtsp-server/rtsp-server.c:
8609 * gst/rtsp-server/rtsp-session-pool.c:
8610 * gst/rtsp-server/rtsp-session.c:
8611 rtsp: allow NULL func in filters
8612 Passing a null function make the filters return a list of
8615 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8617 * gst/rtsp-server/rtsp-address-pool.c:
8618 * tests/check/gst/addresspool.c:
8619 address-pool: fix address increment
8620 Use a guint instead of guint8 to increment the address. It's still not
8621 completely correct because a guint might not be able to hold the complete
8622 address range, but that's an enhacement for later.
8623 Add unit test to test improved behaviour.
8624 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8626 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8628 * gst/rtsp-server/rtsp-client.c:
8629 * tests/check/gst/client.c:
8630 client: allow absolute path in requests
8631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8633 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8635 * gst/rtsp-server/rtsp-client.c:
8636 * gst/rtsp-server/rtsp-client.h:
8637 client: make make_path_from_uri a vmethod
8639 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8641 * docs/libs/gst-rtsp-server-sections.txt:
8642 * gst/rtsp-server/rtsp-stream.c:
8643 * gst/rtsp-server/rtsp-stream.h:
8644 * tests/check/Makefile.am:
8645 * tests/check/gst/stream.c:
8646 stream: Add functions to get rtp and rtcp sockets
8647 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8649 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8651 * gst/rtsp-server/rtsp-context.c:
8652 * gst/rtsp-server/rtsp-context.h:
8653 context: defing a GType for the context
8654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8656 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8658 * gst/rtsp-server/Makefile.am:
8659 * gst/rtsp-server/rtsp-auth.c:
8660 * gst/rtsp-server/rtsp-context.c:
8661 * gst/rtsp-server/rtsp-media.c:
8662 * gst/rtsp-server/rtsp-mount-points.c:
8663 * gst/rtsp-server/rtsp-server.h:
8664 * gst/rtsp-server/rtsp-session-media.c:
8665 * gst/rtsp-server/rtsp-session.c:
8666 * gst/rtsp-server/rtsp-stream.c:
8667 Fixed several GIR warnings
8669 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8671 * gst/rtsp-server/rtsp-auth.c:
8674 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8676 * tests/check/Makefile.am:
8677 * tests/check/gst/token.c:
8678 tests: Add unit tests for token
8679 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8681 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8683 * gst/rtsp-server/rtsp-token.c:
8684 token: Validate args for gst_rtsp_token_is_allowed
8685 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8687 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8689 * gst/rtsp-server/rtsp-token.c:
8690 token: Fix bug when creating empty token
8691 We always want to have a valid GstStructure in the token.
8692 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8694 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8696 * gst/rtsp-server/rtsp-thread-pool.c:
8697 thread-pool: avoid race in shutdown
8698 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8699 don't actually stop the mainloop ever. Solve this race by adding an idle source
8700 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8701 if quit was called before we started it.
8703 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8705 * tests/check/Makefile.am:
8706 * tests/check/gst/permissions.c:
8707 tests: Add unit tests for permissions
8708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8710 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8712 * tests/check/gst/mediafactory.c:
8713 tests: Test mediafactory permissions
8714 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8716 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8718 * gst/rtsp-server/rtsp-permissions.c:
8719 permissions: Fix refcounting when adding/removing roles
8720 Previously a role that was removed was unreffed twice, and when
8721 replacing an existing role the replaced role was freed while still being
8722 referenced. Both bugs are now fixed.
8723 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8725 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8727 * tests/check/gst/media.c:
8728 * tests/check/gst/mediafactory.c:
8729 * tests/check/gst/rtspserver.c:
8730 tests: Check gst_rtsp_url_parse return value
8731 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8733 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8736 Automatic update of common submodule
8737 From 865aa20 to dbedaa0
8739 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8741 * gst/rtsp-server/rtsp-server.c:
8742 rtsp-server: Fix socket leak
8743 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8745 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8747 * gst/rtsp-server/rtsp-session-pool.c:
8748 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8749 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8751 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8753 * examples/.gitignore:
8754 * examples/test-video.c:
8755 examples: fix compilation when WITH_AUTH is defined
8756 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8758 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8761 gitignore: Add new test binary
8763 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8765 * tests/check/Makefile.am:
8766 * tests/check/gst/threadpool.c:
8767 thread-pool: Add unit test for the thread pools
8768 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8770 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8772 * gst/rtsp-server/rtsp-thread-pool.c:
8773 thread-pool: Fix thread leak when reusing threads
8774 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8776 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8778 * gst/rtsp-server/rtsp-server.c:
8779 * tests/check/gst/rtspserver.c:
8780 tests: fixed racy behavior in rtspserver tests
8781 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8783 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8785 * tests/check/gst/addresspool.c:
8786 tests: Improve address pool unit tests
8787 Add a range with mixed IPV4 and IPV6 addresses to pool.
8788 Get an IPV4 address from an IPV6-only pool.
8789 Get an IPV6 address from an IPV4-only pool.
8790 Reserve a IPV6 address from an IPV4-only pool.
8791 Check for unicast addresses in multicast-only pool.
8792 Check for unicast addresses in uni-/multicast-mixed pool.
8793 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8795 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8797 * gst/rtsp-server/rtsp-client.c:
8798 client: append query string in PAUSE/PLAY/TEARDOWN as well
8800 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8802 * gst/rtsp-server/rtsp-client.c:
8803 client: Add query to control path
8804 If the SETUP url contains a query it must be appended to the control
8805 path so that it matches any already created stream in the media. The
8806 query will also be appended to the session media path.
8808 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8810 * gst/rtsp-server/rtsp-media.c:
8811 rtsp-media: remove old line
8813 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8815 * gst/rtsp-server/rtsp-stream.c:
8816 stream: Correct control comparison
8817 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8819 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8821 * gst/rtsp-server/rtsp-media.c:
8822 media: Check dynamically if the pipeline supports seeking
8823 We should not depend on whether or not the pipeline state change
8824 returned NO_PREROLL or not. A media could dynamically change its
8825 element and switch from seekable to non seekable so it's best to test
8826 the seekable nature of the pipeline dynamically when we try to do a seek.
8828 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8830 * gst/rtsp-server/rtsp-media.c:
8831 media: Return FALSE if seeking is not supported
8833 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8835 * gst/rtsp-server/rtsp-media.c:
8836 rtsp-media: don't seek accurate by default
8837 Accurate seeking is perhaps a little overkill in the most common situation and
8838 causes some formats (mp3) over slow media to seek extremely slowly.
8840 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8842 * tests/check/gst/rtspserver.c:
8843 tests: fix unit test
8844 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8846 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8848 * gst/rtsp-server/rtsp-client.c:
8849 client: Reply 400 if media cannot be constructed
8850 Reply 400 Bad Request instead of 503 Service Unavailable if media
8851 cannot be constructed in SETUP.
8852 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8854 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8856 * gst/rtsp-server/rtsp-client.c:
8857 client: Send setup reply once only
8858 If find_media() failed in handle_setup_request() two replies was sent.
8859 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8861 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8864 Automatic update of common submodule
8865 From 6b03ba7 to 865aa20
8867 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8869 * gst/rtsp-server/rtsp-server.c:
8870 server: Emit client-connected signal earlier
8871 Emit client-connected before the client ref is given to a GSource,
8872 otherwise client-connected can be emitted after the client object has
8875 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8877 * gst/rtsp-server/rtsp-address-pool.c:
8878 * gst/rtsp-server/rtsp-address-pool.h:
8879 * gst/rtsp-server/rtsp-stream.c:
8880 * tests/check/gst/addresspool.c:
8881 addresspool: return reason of failure
8882 Let gst_rtsp_address_pool_reserve_address() return the reason why
8883 the address could not be reserved.
8884 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8886 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8889 autogen.sh: Sync behaviour with other GStreamer modules
8890 Allows building from outside of tree amongst other things
8892 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8895 Automatic update of common submodule
8896 From b613661 to 6b03ba7
8898 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8901 Automatic update of common submodule
8902 From 74a6857 to b613661
8904 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8907 Automatic update of common submodule
8908 From 01a7a46 to 74a6857
8910 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8912 * gst/rtsp-server/rtsp-client.c:
8913 client: Do not read beyond end of path string
8914 If the setup was done without a control url, make sure we don't try to read the
8915 non-existing control string and crash.
8917 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-client.c:
8920 client: Fix RTPInfo header
8921 Refactor the method to make the content_base.
8922 Use the content-base and the control url to construct the RTPInfo
8925 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8927 * gst/rtsp-server/rtsp-client.c:
8928 client: map url to path only in describe
8929 Only map the request url to a path in the DESCRIBE method. The SDP then
8930 contains the base and control urls that should be used to SETUP/PAUSE/
8931 PLAY/TEARDOWN the media.
8933 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8935 * gst/rtsp-server/rtsp-client.c:
8936 Revert "client: map URL to path in requests"
8937 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8938 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8939 contains the base and control urls which are used in the SETUP, PLAY,
8940 PAUSE and TEARDOWN requests.
8942 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8944 * gst/rtsp-server/rtsp-client.c:
8945 client: map URL to path in requests
8947 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8949 * gst/rtsp-server/rtsp-client.c:
8950 * gst/rtsp-server/rtsp-mount-points.c:
8951 * gst/rtsp-server/rtsp-mount-points.h:
8952 mount-points: make vmethod to make path from uri
8953 Make a vmethod to transform an url into a path. The path is then used to lookup
8954 the factory. This makes it possible to also use other bits of the url, such as
8955 the query parameters, to locate the factory.
8957 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8959 * gst/rtsp-server/rtsp-thread-pool.c:
8960 * gst/rtsp-server/rtsp-thread-pool.h:
8961 thread-pool: Add cleanup to wait for the threadpool to finish
8962 Also fix race condition if two threads are asking for the first
8963 thread from the thread pool at once. This would case two internal
8964 GThreadPools to be created.
8965 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8967 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8969 * gst/rtsp-server/rtsp-client.c:
8970 * tests/check/gst/client.c:
8971 client: free threadpool
8972 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8974 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8976 * tests/check/gst/mountpoints.c:
8977 mountpoints tests: unref matched factories
8978 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8980 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8982 * tests/check/gst/media.c:
8983 media tests: unref thread pool and caps
8984 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8986 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8988 * gst/rtsp-server/rtsp-auth.c:
8989 * gst/rtsp-server/rtsp-media-factory.c:
8990 * gst/rtsp-server/rtsp-media.c:
8991 auth, media, media-factory: unref permissions
8992 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8994 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8996 * examples/Makefile.am:
8997 Makefile: add rule for appsrc example
8999 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * examples/test-appsrc.c:
9002 tests: add appsrc example
9003 Add an example on how to use appsrc to feed the server pipeline with data.
9005 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
9007 * gst/rtsp-server/rtsp-client.c:
9008 rtsp-client: remove query part from content-base string
9009 Make sure that after the control url has been resolved, it's
9010 not a part of the query-string.
9011 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
9013 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9015 * gst/rtsp-server/rtsp-client.c:
9016 client: don't check url in response
9017 There is no url or method in the response to check
9019 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9021 * gst/rtsp-server/rtsp-client.c:
9022 * gst/rtsp-server/rtsp-client.h:
9023 Add handle-response signal for when we receive a GET_PARAMETER response
9025 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9027 * gst/rtsp-server/rtsp-server.c:
9028 Fix gst_rtsp_server_client_filter, using wrong variable type
9030 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
9032 * gst/rtsp-server/rtsp-media-factory-uri.c:
9033 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
9034 For AAC we need to check for framed=true instead of parsed=true.
9035 https://bugzilla.gnome.org/show_bug.cgi?id=701384
9037 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9039 * gst/rtsp-server/rtsp-stream.c:
9040 stream: optimize pipeline for protocols
9041 When TCP is not an allowed protocol for the stream, avoid creating the
9042 appsrc/appsink/queue and tee elements.
9044 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * gst/rtsp-server/rtsp-media.c:
9047 media: set protocols on streams
9049 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9051 * gst/rtsp-server/rtsp-client.c:
9052 client: use protocols supported by stream
9054 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9056 * gst/rtsp-server/rtsp-media-factory.c:
9057 * gst/rtsp-server/rtsp-media.c:
9058 * gst/rtsp-server/rtsp-stream.c:
9059 media-factory: allow all protocols
9061 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9063 * gst/rtsp-server/rtsp-media.c:
9064 media: configure protocols in new streams
9066 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9068 * gst/rtsp-server/rtsp-stream.c:
9069 * gst/rtsp-server/rtsp-stream.h:
9070 stream: add protocols property
9072 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9074 * gst/rtsp-server/rtsp-media.c:
9075 rtsp-media: send state in "new-state" signal
9076 https://bugzilla.gnome.org/show_bug.cgi?id=705110
9078 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
9081 build: add subdir-objects to AM_INIT_AUTOMAKE
9082 Fixes warnings with automake 1.14
9083 https://bugzilla.gnome.org/show_bug.cgi?id=705350
9085 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9087 * docs/libs/gst-rtsp-server-sections.txt:
9088 * gst/rtsp-server/rtsp-client.c:
9089 * gst/rtsp-server/rtsp-server.c:
9090 * gst/rtsp-server/rtsp-server.h:
9091 server: add method to iterate clients of server
9093 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9095 * gst/rtsp-server/rtsp-media.c:
9096 * gst/rtsp-server/rtsp-media.h:
9097 Add vmethod for rtsp-media subclass to access rtpbin
9099 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9101 * gst/rtsp-server/rtsp-client.h:
9102 small documentation fix
9104 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9106 * gst/rtsp-server/rtsp-client.c:
9107 Do not take range header if range is invalid
9109 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9111 * docs/libs/gst-rtsp-server-sections.txt:
9112 * gst/rtsp-server/rtsp-media.c:
9113 media: add docs for new method
9115 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9117 * gst/rtsp-server/rtsp-media.c:
9118 * gst/rtsp-server/rtsp-media.h:
9119 Add API to rtsp-media set the pipeline's state
9121 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9123 * gst/rtsp-server/rtsp-media.c:
9124 Update current position/duration when gst_rtsp_media_get_range_string is called
9126 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * examples/test-cgroups.c:
9129 tests: add some more docs
9131 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9133 * examples/test-cgroups.c:
9134 * gst/rtsp-server/Makefile.am:
9135 * gst/rtsp-server/rtsp-auth.c:
9136 * gst/rtsp-server/rtsp-auth.h:
9137 * gst/rtsp-server/rtsp-client.c:
9138 * gst/rtsp-server/rtsp-client.h:
9139 * gst/rtsp-server/rtsp-context.c:
9140 * gst/rtsp-server/rtsp-context.h:
9141 * gst/rtsp-server/rtsp-params.c:
9142 * gst/rtsp-server/rtsp-params.h:
9143 * gst/rtsp-server/rtsp-server.c:
9144 * gst/rtsp-server/rtsp-thread-pool.c:
9145 * gst/rtsp-server/rtsp-thread-pool.h:
9146 * tests/check/gst/client.c:
9147 ClientState -> Context
9148 Rename the clientstate to context and put the code in a separate file.
9150 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9152 * examples/test-auth.c:
9153 * gst/rtsp-server/rtsp-auth.c:
9154 * gst/rtsp-server/rtsp-auth.h:
9155 auth: add support for default token
9156 The default token is used when the user is not authenticated and can be used to
9157 give minimal permissions.
9159 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9161 * examples/test-auth.c:
9162 * gst/rtsp-server/rtsp-auth.c:
9163 auth: use defines when possible
9165 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9167 * gst/rtsp-server/rtsp-address-pool.c:
9168 address-pool: improve docs
9170 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9172 * gst/rtsp-server/rtsp-permissions.c:
9173 permissions: add the role to the copy
9175 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
9177 * gst/rtsp-server/rtsp-permissions.c:
9178 permissions: Also copy the roles
9180 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
9182 * gst/rtsp-server/rtsp-permissions.c:
9183 permissions: Make it build
9185 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9187 * gst/rtsp-server/rtsp-address-pool.h:
9190 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9192 * docs/libs/gst-rtsp-server-sections.txt:
9193 * gst/rtsp-server/rtsp-auth.c:
9194 * gst/rtsp-server/rtsp-auth.h:
9195 * gst/rtsp-server/rtsp-media.c:
9196 * gst/rtsp-server/rtsp-session-media.c:
9197 * gst/rtsp-server/rtsp-stream-transport.c:
9198 * gst/rtsp-server/rtsp-stream-transport.h:
9199 * gst/rtsp-server/rtsp-stream.c:
9200 * tests/check/gst/client.c:
9203 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9205 * docs/libs/gst-rtsp-server-sections.txt:
9206 * gst/rtsp-server/rtsp-address-pool.c:
9207 * gst/rtsp-server/rtsp-address-pool.h:
9208 * tests/check/gst/addresspool.c:
9209 * tests/check/gst/rtspserver.c:
9210 address-pool: cleanups
9211 Remove redundant method, improve docs.
9213 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9215 * docs/libs/gst-rtsp-server-sections.txt:
9216 * gst/rtsp-server/rtsp-auth.h:
9217 * gst/rtsp-server/rtsp-permissions.c:
9218 * gst/rtsp-server/rtsp-permissions.h:
9219 * gst/rtsp-server/rtsp-token.c:
9222 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9224 * gst/rtsp-server/rtsp-permissions.c:
9225 permissions: implement _remove_role
9227 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9229 * gst/rtsp-server/rtsp-permissions.c:
9230 permissions: update docs
9232 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9234 * tests/check/gst/client.c:
9235 tests: simplify tests
9236 Client settings are now disabled by default so we don't need an auth
9237 module to disable them.
9239 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9241 * gst/rtsp-server/rtsp-auth.c:
9242 auth: add default authorizations
9243 When no auth module is specified, use our table of defaults to look up the
9244 default value of the check instead of always allowing everything. This was
9245 we can disallow client settings by default.
9247 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9250 README: update readme
9252 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9254 * gst/rtsp-server/rtsp-thread-pool.c:
9255 * gst/rtsp-server/rtsp-thread-pool.h:
9256 thread-pool: add more docs
9258 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9260 * gst/rtsp-server/rtsp-thread-pool.c:
9261 * gst/rtsp-server/rtsp-thread-pool.h:
9262 thread-pool: fix race in thread reuse
9263 If we try to reuse a thread right after we made it stop, we end up using a
9264 stopped thread. Catch this case and only reuse threads that are not stopping.
9266 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9268 * gst/rtsp-server/rtsp-server.c:
9269 server: add small debug
9271 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9273 * tests/check/gst/client.c:
9275 Add some permissions to media so we can use the auth and enable
9278 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9280 * gst/rtsp-server/rtsp-client.c:
9281 client: support pushed context in handle_request
9282 If we already have a pushed state, reuse it and add our own things. This makes
9283 it easier to write tests.
9285 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9287 * gst/rtsp-server/rtsp-auth.c:
9288 auth: don't auth on methods
9289 Don't authorize on methods anymore but on the resources that we
9290 try to access, this is more flexible.
9291 Move the authorization checks to where they are needed and let the
9292 check return the response on error.
9294 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9296 * gst/rtsp-server/rtsp-mount-points.c:
9297 mount-points: add some debug
9299 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9301 * tests/check/gst/client.c:
9302 tests: almost fix test
9304 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9306 * gst/rtsp-server/rtsp-auth.c:
9307 * gst/rtsp-server/rtsp-auth.h:
9308 * gst/rtsp-server/rtsp-client.c:
9309 * gst/rtsp-server/rtsp-client.h:
9310 * gst/rtsp-server/rtsp-server.c:
9311 * gst/rtsp-server/rtsp-server.h:
9312 auth: let the auth module check client_settings
9313 Let the auth module decide if client settings are allowed for the
9316 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9318 * gst/rtsp-server/rtsp-token.c:
9319 * gst/rtsp-server/rtsp-token.h:
9320 token: add method to check boolean permission
9322 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9324 * examples/test-auth.c:
9325 * examples/test-cgroups.c:
9326 * gst/rtsp-server/rtsp-token.c:
9327 * gst/rtsp-server/rtsp-token.h:
9328 token: simplify token constructor
9329 Use variable arguments to make easier API.
9331 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9333 * examples/test-auth.c:
9334 * examples/test-cgroups.c:
9335 * gst/rtsp-server/rtsp-media-factory.c:
9336 * gst/rtsp-server/rtsp-media-factory.h:
9337 media-factory: add convenience API for factory
9339 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9341 * examples/test-auth.c:
9342 * examples/test-cgroups.c:
9343 * gst/rtsp-server/rtsp-permissions.c:
9344 * gst/rtsp-server/rtsp-permissions.h:
9345 permissions: simplify API a little
9346 Avoid passing GstStructure in the add_role method, use varargs instead
9347 to construct the structure behind the scenes. We can then also use the
9348 structure name as the role and simplify some more logic.
9350 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9352 * gst/rtsp-server/rtsp-auth.c:
9355 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9357 * gst/rtsp-server/rtsp-auth.c:
9358 * gst/rtsp-server/rtsp-auth.h:
9359 * gst/rtsp-server/rtsp-client.c:
9360 auth: handle unauthorized response
9361 Move handling of the unauthorized response to the auth module, it can add
9362 the appropriate headers to request authorization for the required method
9363 much better than the client.
9365 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9367 * gst/rtsp-server/rtsp-client.c:
9368 * gst/rtsp-server/rtsp-client.h:
9369 client: allow for sending any message, not only requests
9370 Change the _send_request() method to _send_message() so that we
9371 can both send requests and replies.
9373 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9375 * docs/libs/gst-rtsp-server-sections.txt:
9376 * gst/rtsp-server/rtsp-server.h:
9379 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9381 * examples/test-video.c:
9382 * gst/rtsp-server/rtsp-auth.c:
9383 * gst/rtsp-server/rtsp-auth.h:
9384 * gst/rtsp-server/rtsp-server.c:
9385 * gst/rtsp-server/rtsp-server.h:
9386 auth: move TLS handling to auth module
9387 Remove the TLS settings on the server and move it to the auth module because
9388 that is where security related bits go.
9390 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9392 * gst/rtsp-server/rtsp-client.c:
9393 * gst/rtsp-server/rtsp-client.h:
9394 client: add state push/pop
9396 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9398 * gst/rtsp-server/rtsp-client.c:
9399 * gst/rtsp-server/rtsp-client.h:
9400 client: add connection to state
9402 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9404 * gst/rtsp-server/rtsp-mount-points.c:
9405 mount-points: fix debug
9407 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9409 * tests/check/gst/media.c:
9410 tests: fix media test
9412 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9414 * gst/rtsp-server/rtsp-thread-pool.c:
9415 thread-pool: we don't require a state
9417 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9419 * gst/rtsp-server/rtsp-server.c:
9420 server: let context ref the server
9421 So that we don't risk losing the server object early anc crash.
9423 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9425 * tests/check/gst/client.c:
9426 tests: fix client test
9428 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9431 * docs/libs/gst-rtsp-server-docs.sgml:
9432 * docs/libs/gst-rtsp-server-sections.txt:
9433 * gst/rtsp-server/rtsp-address-pool.c:
9434 * gst/rtsp-server/rtsp-auth.c:
9435 * gst/rtsp-server/rtsp-client.c:
9436 * gst/rtsp-server/rtsp-client.h:
9437 * gst/rtsp-server/rtsp-media-factory-uri.c:
9438 * gst/rtsp-server/rtsp-media-factory.c:
9439 * gst/rtsp-server/rtsp-media-factory.h:
9440 * gst/rtsp-server/rtsp-media.c:
9441 * gst/rtsp-server/rtsp-mount-points.c:
9442 * gst/rtsp-server/rtsp-params.c:
9443 * gst/rtsp-server/rtsp-permissions.c:
9444 * gst/rtsp-server/rtsp-sdp.c:
9445 * gst/rtsp-server/rtsp-server.c:
9446 * gst/rtsp-server/rtsp-server.h:
9447 * gst/rtsp-server/rtsp-session-media.c:
9448 * gst/rtsp-server/rtsp-session-pool.c:
9449 * gst/rtsp-server/rtsp-session.c:
9450 * gst/rtsp-server/rtsp-stream-transport.c:
9451 * gst/rtsp-server/rtsp-stream.c:
9452 * gst/rtsp-server/rtsp-thread-pool.c:
9453 * gst/rtsp-server/rtsp-token.c:
9456 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9458 * gst/rtsp-server/rtsp-session-pool.c:
9459 * gst/rtsp-server/rtsp-session-pool.h:
9460 session-pool: make vmethod to create a session
9461 Make a vmethod to create a sessions so that subclasses can create
9462 custom session objects
9464 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9466 * gst/rtsp-server/rtsp-auth.c:
9467 * gst/rtsp-server/rtsp-media-factory.h:
9468 * gst/rtsp-server/rtsp-media.h:
9469 * gst/rtsp-server/rtsp-mount-points.h:
9470 * gst/rtsp-server/rtsp-session-pool.h:
9471 * gst/rtsp-server/rtsp-stream.h:
9474 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9476 * docs/libs/gst-rtsp-server-docs.sgml:
9477 * docs/libs/gst-rtsp-server-sections.txt:
9478 * gst/rtsp-server/rtsp-address-pool.c:
9479 * gst/rtsp-server/rtsp-address-pool.h:
9480 * gst/rtsp-server/rtsp-auth.c:
9481 * gst/rtsp-server/rtsp-client.h:
9482 * gst/rtsp-server/rtsp-media-factory.h:
9483 * gst/rtsp-server/rtsp-media.c:
9484 * gst/rtsp-server/rtsp-media.h:
9485 * gst/rtsp-server/rtsp-permissions.c:
9486 * gst/rtsp-server/rtsp-permissions.h:
9487 * gst/rtsp-server/rtsp-server.h:
9488 * gst/rtsp-server/rtsp-session-media.c:
9489 * gst/rtsp-server/rtsp-session-media.h:
9490 * gst/rtsp-server/rtsp-session-pool.h:
9491 * gst/rtsp-server/rtsp-session.h:
9492 * gst/rtsp-server/rtsp-stream-transport.h:
9493 * gst/rtsp-server/rtsp-stream.c:
9494 * gst/rtsp-server/rtsp-thread-pool.h:
9497 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9500 * examples/Makefile.am:
9501 configure: compile cgroup example conditionally
9502 Only compile the cgroup example when we have libcgroup
9504 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9507 * examples/Makefile.am:
9508 * examples/test-cgroups.c:
9509 examples: add cgroups example
9511 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9513 * tests/check/gst/rtspserver.c:
9514 tests: fix compilation
9516 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9518 * gst/rtsp-server/rtsp-thread-pool.c:
9519 thread-pool: fix vmethod invocation
9521 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9523 * gst/rtsp-server/rtsp-thread-pool.c:
9524 * gst/rtsp-server/rtsp-thread-pool.h:
9525 thread-pool: store thread type in thread
9527 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9529 * gst/rtsp-server/rtsp-client.c:
9530 client: pass thread from pool to media _prepare
9531 Get a thread from the configured threadpool and pass it to the prepare method of
9534 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9536 * gst/rtsp-server/rtsp-media.c:
9537 * gst/rtsp-server/rtsp-media.h:
9538 media: Accept a thread in _prepare
9539 Remove out own threadpool handling and use the provided thread and
9540 maincontext for the bus messages and the state changes.
9542 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9544 * gst/rtsp-server/rtsp-server.c:
9545 server: configure client thread pool
9547 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9549 * gst/rtsp-server/rtsp-client.c:
9550 * gst/rtsp-server/rtsp-client.h:
9551 client: add method to configure thread pool
9553 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9555 * gst/rtsp-server/rtsp-client.h:
9556 * gst/rtsp-server/rtsp-server.c:
9557 * gst/rtsp-server/rtsp-server.h:
9558 server: use thread pool
9559 Use the thread pool instead of doing our own thing.
9561 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9563 * gst/rtsp-server/Makefile.am:
9564 * gst/rtsp-server/rtsp-thread-pool.c:
9565 * gst/rtsp-server/rtsp-thread-pool.h:
9566 thread-pool: add object to manage threads
9567 Add an object to manage the client and media threads.
9569 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9571 * gst/rtsp-server/rtsp-auth.c:
9572 auth: debug authorization check
9574 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9576 * gst/rtsp-server/rtsp-media.c:
9577 media: start media pipeline in context
9578 Start the media pipeline in the provided context (or our default one
9579 when NULL). This makes sure that we run the bus thread in this context and that
9580 all media threads are children of this context.
9582 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9584 * gst/rtsp-server/rtsp-media-factory.c:
9585 factory: pass permissions to media by default
9587 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9589 * examples/test-auth.c:
9590 test: add permissions to auth test
9591 Ass some permissions to the media factory in the test.
9593 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9595 * gst/rtsp-server/rtsp-auth.c:
9596 * gst/rtsp-server/rtsp-auth.h:
9597 * gst/rtsp-server/rtsp-client.c:
9598 auth: simplify auth checks
9599 Remove client from methods, it's now in the state
9600 Perform the check specified by the string, use the information from the
9601 thread local context.
9603 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9605 * gst/rtsp-server/rtsp-client.c:
9606 * gst/rtsp-server/rtsp-client.h:
9607 client: add state to current thread
9608 Add the client to the ClientState object.
9609 Place the ClientState on the current thread.
9611 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9613 * gst/rtsp-server/rtsp-media-factory.c:
9614 * gst/rtsp-server/rtsp-media-factory.h:
9615 * gst/rtsp-server/rtsp-media.c:
9616 * gst/rtsp-server/rtsp-media.h:
9617 media: make it possible to set permissions
9618 Make it possible to set permissions on media and media factory objects
9620 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9622 * gst/rtsp-server/Makefile.am:
9623 * gst/rtsp-server/rtsp-permissions.c:
9624 * gst/rtsp-server/rtsp-permissions.h:
9625 permissions: add permissions object
9626 Add a mini object to store permissions based on a role.
9628 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9630 * examples/test-auth.c:
9631 * gst/rtsp-server/rtsp-auth.c:
9632 * gst/rtsp-server/rtsp-auth.h:
9633 * gst/rtsp-server/rtsp-client.c:
9634 auth: add auth checks
9635 Add an enum with auth checks and implement the checks in the auth object.
9636 Perform the checks from the client.
9638 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9640 * examples/test-auth.c:
9641 * gst/rtsp-server/rtsp-auth.c:
9642 * gst/rtsp-server/rtsp-auth.h:
9643 * gst/rtsp-server/rtsp-client.h:
9644 auth: use the token after authentication
9645 After we authenticated a user, keep the Token around in the state.
9647 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9649 * gst/rtsp-server/rtsp-client.c:
9650 * gst/rtsp-server/rtsp-media.c:
9651 * gst/rtsp-server/rtsp-media.h:
9652 * tests/check/gst/media.c:
9653 media: add optional context for bus messages
9654 Add an optional mainloop to _prepare that will handle the bus messages instead
9655 of always using the shared mainloop.
9657 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9659 * gst/rtsp-server/Makefile.am:
9660 * gst/rtsp-server/rtsp-token.c:
9661 * gst/rtsp-server/rtsp-token.h:
9662 token: add authorization token
9663 Add a simply miniobject that contains the authorizations. The object contains a
9664 GstStructure that hold all authorization fields. When a user is authenticated,
9665 the auth module will create a Token for the user. The token is then used to
9666 check what operations the user is allowed to do and various other configuration
9669 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * examples/test-auth.c:
9672 * gst/rtsp-server/rtsp-auth.c:
9673 * gst/rtsp-server/rtsp-auth.h:
9674 * gst/rtsp-server/rtsp-client.c:
9675 * gst/rtsp-server/rtsp-client.h:
9676 * gst/rtsp-server/rtsp-media-factory.c:
9677 * gst/rtsp-server/rtsp-media-factory.h:
9678 * gst/rtsp-server/rtsp-media.c:
9679 * gst/rtsp-server/rtsp-media.h:
9680 auth: remove auth from media and factory
9681 Remove the auth object from media and factory. We want to have the RTSPClient
9682 authenticate and authorize resources, there is no need to place another auth
9683 manager on the media/factory.
9685 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9687 * examples/test-auth.c:
9688 * gst/rtsp-server/rtsp-auth.c:
9689 * gst/rtsp-server/rtsp-auth.h:
9690 * gst/rtsp-server/rtsp-client.h:
9691 auth: add support for multiple basic auth tokens
9692 Make it possible to add multiple basic authorisation tokens to one authorization
9693 object. Associate with each token an authorization group that will define what
9694 capabilities are allowed.
9696 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9698 * gst/rtsp-server/rtsp-client.c:
9699 client: error out on non-aggregate control
9700 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9702 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9704 * gst/rtsp-server/rtsp-client.c:
9705 client: rework setup request a little
9706 Cache the media in DESCRIBE based on the longest matching path with the uri
9707 that we can find in the mount points.
9708 Rework the setup request a little to get the media from the session or from
9709 the longest matching path, this way we can derive the control string as
9710 everything after the path instead of hardcoding it.
9711 Find the stream based on the control string and only open a session when all
9714 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9716 * gst/rtsp-server/rtsp-media.c:
9717 * gst/rtsp-server/rtsp-media.h:
9718 media: add method to find a stream by control url
9720 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9722 * gst/rtsp-server/rtsp-stream.c:
9723 * gst/rtsp-server/rtsp-stream.h:
9724 stream: add method to check control url of stream
9726 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9728 * gst/rtsp-server/rtsp-client.c:
9729 * gst/rtsp-server/rtsp-session-media.c:
9730 * gst/rtsp-server/rtsp-session-media.h:
9731 * gst/rtsp-server/rtsp-session.c:
9732 * gst/rtsp-server/rtsp-session.h:
9733 session: use path matching for session media
9734 Use a path string instead of a uri to lookup session media in the sessions. Also
9735 use path matching to find the largest possible path that matches.
9737 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9739 * gst/rtsp-server/rtsp-client.c:
9740 * gst/rtsp-server/rtsp-mount-points.c:
9741 * gst/rtsp-server/rtsp-mount-points.h:
9742 * tests/check/gst/mountpoints.c:
9743 mount-points: remove useless vmethod
9744 Making lookups in the mount points should not be done with a URL, if there is a
9745 mapping to be done from URL to mount points, we'll need to do it somewhere
9748 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9750 * gst/rtsp-server/rtsp-mount-points.c:
9751 * gst/rtsp-server/rtsp-mount-points.h:
9752 * tests/check/gst/mountpoints.c:
9753 mount-points: improve mount point searching
9754 Use a GSequence to keep track of the mount points.
9755 Match a URL to the longest matching registered mount point. This should be the
9756 URL to perform aggreagate control and the remainder is the stream specific
9758 Add some unit tests for this.
9760 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9762 * gst/rtsp-server/Makefile.am:
9763 rtsp-server: Allow building of static library
9765 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9767 * tests/check/gst/mediafactory.c:
9768 tests: fix compilation
9770 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9772 * gst/rtsp-server/rtsp-sdp.c:
9773 sdp: get control string from stream
9774 Use the control string as configured in the stream.
9776 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9778 * gst/rtsp-server/rtsp-stream.c:
9779 * gst/rtsp-server/rtsp-stream.h:
9780 stream: add methods and property to set control string
9782 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9784 * gst/rtsp-server/rtsp-client.c:
9786 Rename variables for clarity
9787 Keep media in state when we can
9789 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9791 * gst/rtsp-server/rtsp-client.c:
9792 * gst/rtsp-server/rtsp-stream.c:
9793 * gst/rtsp-server/rtsp-stream.h:
9794 stream: add more support for IPv6
9795 Rename _get_address to _get_multicast_address in GstRTSPStream to
9796 make it clear that this function only deals with multicast.
9797 Make it possible to have both an IPv4 and IPv6 multicast address on
9798 a stream. Give the client an IPv4 or IPv6 address depending on the
9799 address it used to connect to the server.
9800 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9802 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9804 * gst/rtsp-server/rtsp-client.c:
9807 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9809 * gst/rtsp-server/rtsp-stream.c:
9810 stream: handle failed port allocation
9811 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9812 can't allocate any family at all. Also keep track of what port families we
9814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9816 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9818 * gst/rtsp-server/rtsp-stream.c:
9819 stream: improve docs
9821 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9823 * gst/rtsp-server/rtsp-stream-transport.c:
9824 stream-transport: remove old if 0 block
9826 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9828 * tests/check/gst/client.c:
9830 gst_rtsp_client_get_uri() has been removed
9831 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9833 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9835 * gst/rtsp-server/rtsp-client.c:
9836 * gst/rtsp-server/rtsp-client.h:
9837 client: add method to filter managed sessions
9838 Add a method to filter the sessions managed by this client connection.
9839 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9841 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9843 * gst/rtsp-server/rtsp-client.c:
9844 * gst/rtsp-server/rtsp-client.h:
9845 client: remove _get_uri() method
9846 Remove the get_uri() method on the client. A client has no uri, the uri
9847 property is an internal property to manage the last cached media for
9850 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9852 * gst/rtsp-server/rtsp-media-factory.h:
9853 media-factory: fix typo
9855 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9857 * gst/rtsp-server/rtsp-media.c:
9858 rtsp-media: Do not leak the query in default_query_stop
9859 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9861 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9863 * gst/rtsp-server/rtsp-media.c:
9864 media: don't unlock when conversion fails
9865 Don't unlock the state lock when conversion fails because it was not locked.
9867 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9869 * gst/rtsp-server/rtsp-media.c:
9870 * gst/rtsp-server/rtsp-media.h:
9871 Add query_position and query_stop vmethods to rtsp-media
9873 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9875 * gst/rtsp-server/rtsp-media.c:
9876 Fix typo in property install for rtsp-media's time-provider
9878 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9880 * gst/rtsp-server/rtsp-client.c:
9881 * gst/rtsp-server/rtsp-client.h:
9882 client: clean some variables
9883 Clean some variables and add some guards to _send_request()
9885 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9887 * gst/rtsp-server/rtsp-client.c:
9888 * gst/rtsp-server/rtsp-client.h:
9889 Add gst_rtsp_client_send_request API
9890 This makes it possible to send arbitrary messages to a client, such as
9891 SET_PARAMETER or GET_PARAMETER
9893 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9895 * gst/rtsp-server/rtsp-media.c:
9896 * gst/rtsp-server/rtsp-media.h:
9897 media: add _get_element() method
9898 Add method to get the element used when creating the media.
9899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9901 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9903 * gst/rtsp-server/rtsp-media.c:
9906 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9908 * gst/rtsp-server/rtsp-stream.c:
9909 * gst/rtsp-server/rtsp-stream.h:
9910 stream: allow access to the rtp session
9911 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9913 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9915 * gst/rtsp-server/rtsp-stream.c:
9916 * gst/rtsp-server/rtsp-stream.h:
9917 dscp qos support in gst-rtsp-stream
9918 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9920 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9922 * tests/check/gst/rtspserver.c:
9924 Actually do what the comment says. Also keep the old code around, not sure what
9925 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9926 it currently doesn't.
9928 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9930 * gst/rtsp-server/rtsp-client.c:
9931 client: also watch newly created session
9932 When we newly created a session, start watching it immediately instead of
9933 on the next request.
9935 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9937 * tests/check/gst/client.c:
9938 tests: add unit test for new-session
9939 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9941 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9943 * gst/rtsp-server/rtsp-client.c:
9944 client: emit new-session when new session is created
9945 Only emit new-session when we created a new session for a client, not when a
9946 client picked up a previous session.
9947 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9949 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9951 * gst/rtsp-server/rtsp-client.c:
9952 client: handle asterisk as path in requests
9953 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9955 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9957 * gst/rtsp-server/rtsp-media.c:
9958 media: handle segment query format mismatch
9959 It's possible that the segment query returns with a different format than what
9960 we asked for, handle this case also.
9962 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9964 * gst/rtsp-server/rtsp-media.c:
9965 media: use segment stop in collect_media_stats
9966 Use segment stop instead of duration as range end point.
9967 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9969 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9971 * gst/rtsp-server/rtsp-media.c:
9972 * tests/check/gst/media.c:
9973 rtsp-media: Do not leak the element in take_pipeline
9974 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9976 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9978 * gst/rtsp-server/rtsp-client.c:
9979 * gst/rtsp-server/rtsp-client.h:
9980 rtsp-client: Make configure_client_transport virtual
9981 This patch makes configure_client_transport virtual. The functionality is
9982 needed to handle some weird clients sending multicast transport settings as url
9984 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9986 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9988 * gst/rtsp-server/rtsp-client.c:
9989 * gst/rtsp-server/rtsp-client.h:
9990 rtsp-client: Make param_set and param_get virtual
9991 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9993 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9995 * gst/rtsp-server/rtsp-client.c:
9996 * gst/rtsp-server/rtsp-media.c:
9997 * gst/rtsp-server/rtsp-media.h:
9998 media: convert_range replaces get_range_times
9999 get_range_times worked for handling UTC ranges for seeks, but we also
10000 need to convert back from NPT to the requested unit in
10001 get_range_string. convert_range is now used for both.
10002 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
10004 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10006 * gst/rtsp-server/rtsp-client.c:
10007 * gst/rtsp-server/rtsp-sdp.c:
10008 * gst/rtsp-server/rtsp-sdp.h:
10009 sdp: cleanup sdp info
10010 We don't need to pass the proto, we can more easily check a boolean.
10011 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
10013 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
10015 * gst/rtsp-server/rtsp-sdp.c:
10016 use 0.0.0.0 or :: for c= line instead of server address
10018 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
10020 * gst/rtsp-server/rtsp-client.c:
10021 use local address, not remote, in SDP
10022 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
10024 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10027 Automatic update of common submodule
10028 From 098c0d7 to 01a7a46
10030 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
10032 * gst/rtsp-server/rtsp-media.c:
10033 * gst/rtsp-server/rtsp-media.h:
10034 media: possibility to override range time conversion
10035 Make it possible to override the conversion from GstRTSPTimeRange to
10036 GstClockTimes, that is done before seeking on the media
10037 pipeline. Overriding can be useful for UTC ranges, where the default
10038 conversion gives nanoseconds since 1900.
10039 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
10041 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
10043 * gst/rtsp-server/rtsp-server.c:
10044 * gst/rtsp-server/rtsp-server.h:
10045 rtsp-server: Expose the use_client_settings API
10046 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
10048 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
10050 * gst/rtsp-server/rtsp-client.c:
10051 * gst/rtsp-server/rtsp-stream.c:
10052 * gst/rtsp-server/rtsp-stream.h:
10053 rtspstream: handle both ipv4 and ipv6 clients
10054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
10056 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10058 * gst/rtsp-server/rtsp-sdp.c:
10059 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
10060 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
10061 We already have a way to place extra attributes in the SDP by using a string
10062 property with prefix x- or a- in the caps.
10064 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10066 * gst/rtsp-server/rtsp-sdp.c:
10067 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
10068 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
10069 We already have a way to place extra attributes in the SDP, just make a string
10070 property in the payloader with a- or x- prefix.
10072 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10074 * gst/rtsp-server/rtsp-sdp.c:
10075 rtsp: place a- and x- properties as attributes
10076 application/x-rtp has properties with a- and x- prefixes that should be
10077 placed as attributes in the SDP for the media instead of being added to the
10080 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10082 * examples/Makefile.am:
10083 * examples/test-video.c:
10084 example: add TLS example
10086 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10088 * gst/rtsp-server/rtsp-server.c:
10089 * gst/rtsp-server/rtsp-server.h:
10090 server: add support for TLS
10091 Add methods to set and get a TLS certificate.
10092 Add vmethod to configure a new connection. By default, configure the TLS
10093 certificate in a new connection if needed.
10095 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10097 * gst/rtsp-server/rtsp-server.c:
10098 * gst/rtsp-server/rtsp-server.h:
10099 server: remove accept_client vmethod
10100 This vmethod is not very useful so remove it.
10102 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10104 * gst/rtsp-server/rtsp-server.c:
10105 server: don't crash on NULL GError
10107 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
10109 * gst/rtsp-server/rtsp-session-pool.c:
10110 rtsp-session-pool: corrected session timeout detection
10111 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
10113 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10115 * gst/rtsp-server/rtsp-client.c:
10116 client: improve debug
10118 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10120 * gst/rtsp-server/rtsp-client.c:
10121 * gst/rtsp-server/rtsp-client.h:
10122 * gst/rtsp-server/rtsp-server.c:
10123 server: refactor connection setup
10124 Let the server accept the socket connection and construct a GstRTSPConnection
10125 from it. Remove the code from the client and let the client only deal with
10126 a fully configure GstRTSPConnection object.
10127 We will need this later when the server will configure the connection for
10130 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10132 * gst/rtsp-server/rtsp-stream.c:
10133 stream: keep the transport object alive
10134 Keep the transport object alive while we have it as qdata on the
10137 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
10139 * gst/rtsp-server/rtsp-client.c:
10140 * gst/rtsp-server/rtsp-server.c:
10141 rtsp-server: Do not crash on nmapping of server
10142 * generate error when gst_rtsp_connection_accept fails
10143 * do not stop accepting incoming connections because
10144 accepting a client fails
10145 https://bugzilla.gnome.org/show_bug.cgi?id=701072
10147 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
10149 * gst/rtsp-server/rtsp-client.c:
10150 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
10151 https://bugzilla.gnome.org/show_bug.cgi?id=700953
10153 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
10155 * gst/rtsp-server/rtsp-sdp.c:
10156 rtsp-sdp: Parse framerate caps field and set SDP attribute
10157 The SDP attribute and its format is described in RFC4566.
10158 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10160 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
10162 * gst/rtsp-server/rtsp-sdp.c:
10163 rtsp-sdp: Parse width/height from caps and set SDP attribute
10164 The SDP attribute and its format is described in RFC6064.
10165 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10167 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
10169 * gst/rtsp-server/rtsp-sdp.c:
10170 * tests/check/gst/client.c:
10171 rtsp-sdp: add bandwidth line
10172 https://bugzilla.gnome.org/show_bug.cgi?id=699220
10174 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10177 Automatic update of common submodule
10178 From 5edcd85 to 098c0d7
10180 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10182 * tests/check/gst/media.c:
10183 tests: add dynamic payloader prepare/unprepare check
10185 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10187 * gst/rtsp-server/rtsp-media.c:
10188 media: release lock when removing fakesink
10190 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10192 * gst/rtsp-server/rtsp-stream.c:
10193 stream: set elements to NULL before removing
10194 When removing a stream, set the elements to NULL first. This avoids
10195 element-is-not-in-NULL-state errors when we dispose the elements.
10197 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
10200 Automatic update of common submodule
10201 From 3cb3d3c to 5edcd85
10203 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10205 * gst/rtsp-server/rtsp-media.c:
10206 * gst/rtsp-server/rtsp-media.h:
10207 media: listen to pad-removed signals
10208 Listen to the pad-removed signal and remove the stream associated with the
10210 Add signal to be notified of the removed pad.
10211 Remove the fakesink in unprepare()
10212 Fix signatures of the signal methods
10214 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10216 * examples/test-sdp.c:
10217 tests: add example of reusable pipelines
10219 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
10221 * gst/rtsp-server/rtsp-stream.c:
10222 * gst/rtsp-server/rtsp-stream.h:
10223 stream: add method to get the srcpad
10225 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10227 * tests/check/gst/media.c:
10228 check: add media prepare/unprepare test
10229 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10231 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
10233 * gst/rtsp-server/rtsp-media.c:
10234 media: disconnect from signal handlers in unprepare()
10235 We connected to the pad-added and no-more-pads signals in prepare() so
10236 we need to disconnect from them in unprepare().
10237 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10239 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
10241 * gst/rtsp-server/rtsp-media.c:
10242 media: don't free streams array
10243 Don't free the streams array in the unprepare() method, they were not
10244 added in prepare().
10245 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10247 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
10249 * gst/rtsp-server/rtsp-media.c:
10250 media: don't unref the pipeline in unprepare
10251 Unprepare() should undo what prepare() does. Because the pipeline is
10252 not created in prepare(), we should not unref it in unprepare()
10254 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
10256 * gst/rtsp-server/rtsp-stream.c:
10257 stream: clear session and caps for reuse
10258 Set the session and caps to NULL after unref otherwise we might unref
10260 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10262 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
10264 * gst/rtsp-server/rtsp-client.c:
10265 client: send out teardown signal before tearing down
10266 The advantage is that in the signal handler you get direct access to
10267 information about what streams are about to get torn down (in the
10268 GstRTSPClientState).
10269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
10271 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
10273 * gst/rtsp-server/rtsp-client.c:
10274 * gst/rtsp-server/rtsp-client.h:
10275 client: expose connection
10276 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
10278 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
10281 Automatic update of common submodule
10282 From aed87ae to 3cb3d3c
10284 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10286 * gst/rtsp-server/rtsp-media.c:
10287 * gst/rtsp-server/rtsp-media.h:
10288 * gst/rtsp-server/rtsp-session-media.c:
10289 * gst/rtsp-server/rtsp-session-media.h:
10290 media: add method to get the base_time of the pipeline
10291 Together with a shared clock, this base-time could eventually be sent to
10292 the client so that it can reconstruct the exact running-time of the clock
10295 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10297 * gst/rtsp-server/Makefile.am:
10298 * gst/rtsp-server/rtsp-media.c:
10299 * gst/rtsp-server/rtsp-media.h:
10300 * gst/rtsp-server/rtsp-sdp.c:
10301 media: add GstNetTimeProvider support
10302 Add a property to let the media provide a GstNetTimeProvider for its clock.
10303 Make methods to get the clock and nettimeprovider
10304 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
10305 provider and also the current time of the clock. This should make it possible
10306 for (GStreamer) clients to slave their clock to the server clock.
10308 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
10311 Automatic update of common submodule
10312 From 04c7a1e to aed87ae
10314 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10316 * gst/rtsp-server/rtsp-media.c:
10317 media: wait for buffering to complete
10318 Wait for buffering to complete before changing the state to the target state.
10320 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10322 * gst/rtsp-server/rtsp-media.c:
10323 media: small cleanup
10325 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
10327 * tests/check/gst/rtspserver.c:
10328 tests: remove extra unref in test_setup_non_existing_stream
10329 The unref is not needed anymore, teardown runs without it.
10330 https://bugzilla.gnome.org/show_bug.cgi?id=696542
10332 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
10334 * tests/check/gst/rtspserver.c:
10335 tests: GSocketService cleanup in test_bind_already_in_use
10336 Use g_socket_service_stop so the rtspserver test stops listening for
10337 incoming connections in test_bind_already_in_use.
10338 https://bugzilla.gnome.org/show_bug.cgi?id=696541
10340 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
10342 * gst/rtsp-server/rtsp-media-factory.c:
10343 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
10344 Instead use a GWeakRef which is safe to use
10345 This is a known GLib bug, see:
10346 https://bugzilla.gnome.org/show_bug.cgi?id=667145
10348 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
10350 * gst/rtsp-server/rtsp-client.c:
10351 * gst/rtsp-server/rtsp-media.c:
10352 * gst/rtsp-server/rtsp-media.h:
10353 * gst/rtsp-server/rtsp-sdp.c:
10354 * tests/check/gst/media.c:
10355 * tests/check/gst/rtspserver.c:
10356 rtsp-media/client: Reply to PLAY request with same type of Range
10357 Remember the type of Range from the PLAY request and use the same type for
10360 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
10362 * gst/rtsp-server/rtsp-client.c:
10363 * gst/rtsp-server/rtsp-client.h:
10364 * tests/check/gst/client.c:
10365 rtsp-client: expose uri
10367 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
10369 * tests/check/gst/mediafactory.c:
10370 tests: Hold ref while creating second media
10371 To test if the media aren't shared, make sure we keep the first one while creating a second
10372 otherwise the same memory address may be reused.
10374 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
10377 configure: remove out-of-date comment
10379 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10382 .gitignore: ignore more build files
10384 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10386 * tests/check/Makefile.am:
10387 tests: use right _LIBS variable for gst-plugins-base libs
10389 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10391 * tests/check/Makefile.am:
10392 check: add librtp to libs
10394 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10396 * tests/check/gst/rtspserver.c:
10397 tests: Add test to check selecting a port the server will send from
10399 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10401 * tests/check/gst/rtspserver.c:
10402 tests: Make sure packets are actually received
10404 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10406 * gst/rtsp-server/rtsp-stream.c:
10407 stream: Select unicast address from pool if appropriate
10409 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10411 * gst/rtsp-server/rtsp-stream.c:
10412 stream: Properties are always there in Gst 1.0
10414 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10416 * tests/check/gst/addresspool.c:
10417 tests: Add tests for unicast addresses in pool
10419 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10421 * gst/rtsp-server/rtsp-address-pool.c:
10422 * tests/check/gst/addresspool.c:
10423 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10425 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10427 * docs/libs/gst-rtsp-server-sections.txt:
10428 * gst/rtsp-server/rtsp-address-pool.c:
10429 * gst/rtsp-server/rtsp-address-pool.h:
10430 * gst/rtsp-server/rtsp-stream.c:
10431 * tests/check/gst/addresspool.c:
10432 address-pool: Add unicast addresses
10434 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10437 * gst/rtsp-server/rtsp-server.c:
10438 * tests/check/gst/rtspserver.c:
10439 rtsp-server: Limit the number of threads per server instance
10440 If we exceed the maximum, just round robin the clients over the existing
10443 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10445 * gst/rtsp-server/rtsp-server.c:
10446 rtsp-server: No need to store the GMainContext in the client context
10448 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10450 * tests/check/gst/rtspserver.c:
10451 tests: Add test for client disconnection
10453 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10455 * tests/check/gst/rtspserver.c:
10456 tests: Test client and session timeouts with multiple threads
10458 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10460 * gst/rtsp-server/rtsp-address-pool.c:
10461 * gst/rtsp-server/rtsp-auth.c:
10462 * gst/rtsp-server/rtsp-client.c:
10463 * gst/rtsp-server/rtsp-media-factory-uri.c:
10464 * gst/rtsp-server/rtsp-media-factory.c:
10465 * gst/rtsp-server/rtsp-media.c:
10466 * gst/rtsp-server/rtsp-mount-points.c:
10467 * gst/rtsp-server/rtsp-server.c:
10468 * gst/rtsp-server/rtsp-session-media.c:
10469 * gst/rtsp-server/rtsp-session-pool.c:
10470 * gst/rtsp-server/rtsp-session.c:
10471 Document locking and its order
10473 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10475 * tests/check/gst/rtspserver.c:
10476 tests: Test that slow DESCRIBE don't block other clients
10478 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10480 * tests/check/gst/client.c:
10481 tests: Add tests for client-requested multicast address
10483 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10485 * docs/libs/gst-rtsp-server-sections.txt:
10486 docs: Put the various functions in the right sections
10488 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10490 * docs/libs/gst-rtsp-server-docs.sgml:
10491 * docs/libs/gst-rtsp-server-sections.txt:
10492 * gst/rtsp-server/rtsp-address-pool.c:
10493 * gst/rtsp-server/rtsp-address-pool.h:
10494 docs: Generate docs for GstRTSPAddressPool
10496 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10498 * gst/rtsp-server/rtsp-client.c:
10499 * gst/rtsp-server/rtsp-stream.c:
10500 * gst/rtsp-server/rtsp-stream.h:
10501 client: Check client provided addresses against the address pool
10503 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10505 * gst/rtsp-server/rtsp-address-pool.c:
10506 * gst/rtsp-server/rtsp-address-pool.h:
10507 * tests/check/gst/addresspool.c:
10508 address-pool: Add API to request a specific address from the pool
10509 Also add relevant unit tests.
10511 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10513 * tests/check/gst/mediafactory.c:
10514 tests: Check the passing around of a RTSPAddressPool
10515 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10516 way down to the stream.
10518 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10520 * tests/check/gst/addresspool.c:
10521 tests: Add more tests for the address pool
10523 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10525 * gst/rtsp-server/rtsp-address-pool.c:
10526 address-pool: Fix off by one error
10527 When splitting a port range, the port after a skip is not part of range.
10529 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10532 Automatic update of common submodule
10533 From 2de221c to 04c7a1e
10535 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10538 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10539 AM_CONFIG_HEADER was removed in automake 1.13
10540 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10542 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10545 Automatic update of common submodule
10546 From a942293 to 2de221c
10548 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10550 * gst/rtsp-server/rtsp-client.c:
10551 client: make sure the watch exists while sending data
10552 Protect the send_func with a lock. This allows us to wait for sending
10553 to complete before changing the send_func and user_data. We add an
10554 extra ref to the watch to make sure that it remains valid during
10556 When closing the connection, set the send_func to NULL
10557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10559 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10561 * tests/check/Makefile.am:
10562 tests: use GST_*_1_0 environment variables everywhere
10563 The _1_0 suffixed environment variables override the
10564 non-suffixed ones, so if we're in an environment that
10565 sets the _1_0 suffixed ones, such as jhbuild, we need
10566 to set those to make sure ours actually always get
10569 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10572 Automatic update of common submodule
10573 From acb04d9 to a942293
10575 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10577 * gst/rtsp-server/rtsp-client.c:
10578 rtsp-client: set the client backlog
10579 Set the client backlog to a reasonable default
10581 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10583 * gst/rtsp-server/rtsp-media.c:
10584 rtsp-media: Make the element a constructor parameter
10585 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10587 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10589 * docs/libs/Makefile.am:
10590 docs: Link with gcov library when gcov is enabled
10591 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10593 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10595 * gst/rtsp-server/rtsp-media.c:
10596 media: match prepare with unprepare
10597 Really unprepare when there were an equal amount of prepare calls.
10599 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10601 * gst/rtsp-server/rtsp-media.c:
10602 media: media has to be unprepared in finalize
10603 Because unprepare takes away the last ref on the media.
10605 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * gst/rtsp-server/rtsp-client.c:
10608 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10609 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10610 We can't use the refcount to trigger unprepare because it is the unprepare call
10611 that removes the last refcount after all messages are consumed. What we should
10612 probably do is make a prepared refcount and only unprepare when the refcount
10615 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10617 * gst/rtsp-server/rtsp-media.c:
10618 media: let the source unref the last media ref
10619 the last ref to the media is held by the source so we don't need to add more ref
10620 and unrefs, we simply destroy the media when the source is gone.
10622 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10624 * gst/rtsp-server/rtsp-media.c:
10625 media: improve debug
10627 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10629 * gst/rtsp-server/rtsp-media.c:
10631 Make sure we are in the right state when collecting the position and duration.
10632 Only make ourselves PREPARED when we were previously PREPARING.
10634 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10636 * gst/rtsp-server/rtsp-media.c:
10637 media: use g_object_ref/unref for GObjects
10639 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10641 * gst/rtsp-server/rtsp-client.c:
10642 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10643 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10644 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10645 isn't being used anymore.
10647 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10649 * gst/rtsp-server/rtsp-media.c:
10650 Fix compiler warning
10652 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10654 * gst/rtsp-server/rtsp-media-factory-uri.c:
10655 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10657 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10659 * gst/rtsp-server/rtsp-session-media.h:
10662 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10664 * gst/rtsp-server/rtsp-media.c:
10665 * tests/check/gst/media.c:
10666 media: avoid element leak
10668 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10670 * gst/rtsp-server/rtsp-media.c:
10671 media: require an element in media constructor
10673 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10675 * gst/rtsp-server/rtsp-client.c:
10676 Revert "client: TEARDOWN brings that state to Init again"
10677 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10678 The object is already disposed, there is no point in setting the state.
10680 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10682 * gst/rtsp-server/rtsp-client.c:
10683 client: TEARDOWN brings that state to Init again
10685 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10687 * docs/libs/gst-rtsp-server-sections.txt:
10688 * examples/test-auth.c:
10689 * gst/rtsp-server/rtsp-auth.c:
10690 * gst/rtsp-server/rtsp-auth.h:
10691 * gst/rtsp-server/rtsp-client.c:
10692 * gst/rtsp-server/rtsp-client.h:
10693 * gst/rtsp-server/rtsp-media-factory-uri.c:
10694 * gst/rtsp-server/rtsp-media-factory-uri.h:
10695 * gst/rtsp-server/rtsp-media-factory.c:
10696 * gst/rtsp-server/rtsp-media-factory.h:
10697 * gst/rtsp-server/rtsp-media.c:
10698 * gst/rtsp-server/rtsp-media.h:
10699 * gst/rtsp-server/rtsp-mount-points.c:
10700 * gst/rtsp-server/rtsp-mount-points.h:
10701 * gst/rtsp-server/rtsp-sdp.c:
10702 * gst/rtsp-server/rtsp-server.c:
10703 * gst/rtsp-server/rtsp-server.h:
10704 * gst/rtsp-server/rtsp-session-media.c:
10705 * gst/rtsp-server/rtsp-session-media.h:
10706 * gst/rtsp-server/rtsp-session-pool.c:
10707 * gst/rtsp-server/rtsp-session-pool.h:
10708 * gst/rtsp-server/rtsp-session.c:
10709 * gst/rtsp-server/rtsp-session.h:
10710 * gst/rtsp-server/rtsp-stream-transport.c:
10711 * gst/rtsp-server/rtsp-stream-transport.h:
10712 * gst/rtsp-server/rtsp-stream.c:
10713 * gst/rtsp-server/rtsp-stream.h:
10714 * tests/check/gst/media.c:
10715 rtsp: make object details private
10716 Make all object details private
10717 Add methods to access private bits
10719 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10721 * tests/check/Makefile.am:
10722 * tests/check/gst/media.c:
10723 tests: add media tests
10725 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10727 * gst/rtsp-server/rtsp-media.c:
10728 media: check if prepared for some methods
10729 Check that the media object is prepared before doing seek and getting the
10730 current position etc.
10731 Add some g_return checks.
10733 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10735 * tests/check/Makefile.am:
10736 * tests/check/gst/mediafactory.c:
10737 tests: add mediafactory test
10739 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10741 * gst/rtsp-server/rtsp-stream.c:
10742 stream: improve debug
10744 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10746 * gst/rtsp-server/rtsp-media.c:
10747 * gst/rtsp-server/rtsp-media.h:
10748 media: unref pipeline in finalize to avoid leaking it
10750 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10752 * gst/rtsp-server/rtsp-media-factory-uri.c:
10753 * gst/rtsp-server/rtsp-media.c:
10754 rtsp: use gst_object_unref on GstObjects
10756 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10758 * gst/rtsp-server/rtsp-media-factory.c:
10759 media-factory: require an url
10761 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10763 * examples/test-uri.c:
10764 examples: fix include
10766 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10768 * gst/rtsp-server/rtsp-server.h:
10769 server: remove unused include
10771 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10773 * tests/check/Makefile.am:
10774 * tests/check/gst/mountpoints.c:
10775 tests: add test for mountpoints
10777 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10779 * gst/rtsp-server/rtsp-client.c:
10780 client: fix factory leak
10781 Keep the factory in the state object only for authorization checks and make
10782 sure we unref it on failure. Also don't keep invalid objects in the state
10785 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10787 * gst/rtsp-server/rtsp-mount-points.c:
10788 mounts: add g_return_if guards
10790 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10792 * tests/check/gst/client.c:
10793 tests: add more tests
10795 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10797 * gst/rtsp-server/rtsp-client.c:
10798 client: improve debug
10800 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-client.c:
10803 client: improve debug and fix leaks
10804 Cleanup the uri and session when there is a bad request.
10806 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10811 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10813 * tests/check/gst/client.c:
10814 test: add test for session in options request
10816 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10818 * gst/rtsp-server/rtsp-client.c:
10819 client: use 454 when session can't be found
10820 We should use 454 when a session can't be found because there was no session
10821 pool configured in the server. This is not a server configuration problem
10822 because the server on which the request is done might not be the same one that
10823 will keep the sessions for us and so it does not need to support sessions.
10825 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10827 * gst/rtsp-server/rtsp-client.c:
10828 client: only free connection when there is one
10829 It's possible that the client doesn't have a connection when we try to free it.
10831 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10833 * tests/check/Makefile.am:
10834 * tests/check/gst/client.c:
10835 tests: add unit test for the client object
10837 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10839 * gst/rtsp-server/rtsp-client.c:
10840 client: small cleanup
10842 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10844 * gst/rtsp-server/rtsp-client.h:
10845 client: remove unused include
10847 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10849 * gst/rtsp-server/rtsp-client.c:
10850 client: fix compilation
10852 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10854 * gst/rtsp-server/rtsp-client.c:
10855 client: call destroy without the lock
10857 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10859 * gst/rtsp-server/rtsp-client.c:
10860 * gst/rtsp-server/rtsp-client.h:
10861 client: make the client usable without a socket
10862 Make a method to let the client handle a message and a callback when the client
10863 wants us to send a response message back. This makes it possible to also use the
10864 client object without the sockets, which should make it easier to test.
10866 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10868 * gst/rtsp-server/rtsp-client.c:
10869 * gst/rtsp-server/rtsp-client.h:
10870 client: small cleanup
10872 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10874 * docs/libs/gst-rtsp-server-sections.txt:
10875 * gst/rtsp-server/rtsp-client.c:
10876 * gst/rtsp-server/rtsp-client.h:
10877 * gst/rtsp-server/rtsp-server.c:
10878 client: remove reference to server
10879 We don't need to keep a ref to the server
10881 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10883 * gst/rtsp-server/rtsp-client.c:
10884 * gst/rtsp-server/rtsp-client.h:
10885 client: add locking
10886 Also add some g_return_if()
10888 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10890 * gst/rtsp-server/rtsp-client.c:
10891 client: log more errors
10893 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10895 * gst/rtsp-server/rtsp-client.c:
10896 client: fix compilation
10898 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10900 * gst/rtsp-server/rtsp-client.c:
10901 * gst/rtsp-server/rtsp-client.h:
10902 client: add generic close-after-send support
10903 Add a property to send_response() to close the connection after the response has
10904 been sent to the client.
10906 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10909 * docs/libs/gst-rtsp-server-docs.sgml:
10910 * docs/libs/gst-rtsp-server-sections.txt:
10911 * docs/libs/gst-rtsp-server.types:
10912 * examples/test-auth.c:
10913 * examples/test-launch.c:
10914 * examples/test-mp4.c:
10915 * examples/test-multicast.c:
10916 * examples/test-multicast2.c:
10917 * examples/test-ogg.c:
10918 * examples/test-readme.c:
10919 * examples/test-sdp.c:
10920 * examples/test-uri.c:
10921 * examples/test-video.c:
10922 * gst/rtsp-server/Makefile.am:
10923 * gst/rtsp-server/rtsp-auth.h:
10924 * gst/rtsp-server/rtsp-client.c:
10925 * gst/rtsp-server/rtsp-client.h:
10926 * gst/rtsp-server/rtsp-media-mapping.c:
10927 * gst/rtsp-server/rtsp-media-mapping.h:
10928 * gst/rtsp-server/rtsp-mount-points.c:
10929 * gst/rtsp-server/rtsp-mount-points.h:
10930 * gst/rtsp-server/rtsp-server.c:
10931 * gst/rtsp-server/rtsp-server.h:
10932 * gst/rtsp-server/rtsp-session-media.c:
10933 * gst/rtsp-server/rtsp-session-pool.c:
10934 * gst/rtsp-server/rtsp-session-pool.h:
10935 * tests/check/gst/rtspserver.c:
10936 MediaMapping -> MountPoints
10937 Describes better what the object manages.
10939 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10942 configure: bump required version of -base
10944 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10946 * gst/rtsp-server/rtsp-media.c:
10949 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10951 * gst/rtsp-server/rtsp-media.c:
10952 * gst/rtsp-server/rtsp-media.h:
10953 media: support more Range formats
10954 Use the new -base methods to convert the Range string into a seek start and stop
10957 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10959 * examples/test-launch.c:
10960 examples: fix whitespace
10962 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10964 * examples/test-auth.c:
10965 test-auth: add example of how to remove sessions
10966 Add an example of the session filter api.
10968 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10970 * examples/test-uri.c:
10971 test-uri: remove mapping example
10973 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10975 * examples/test-uri.c:
10976 test-uri: fix callback signature
10978 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10980 * gst/rtsp-server/rtsp-media-factory.c:
10981 factory: keep ref to factory while media active
10982 While the media from a factory is alive, keep a ref to the factory.
10983 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10985 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10987 * gst/rtsp-server/rtsp-media-factory-uri.c:
10988 factory-uri: add some debug
10990 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10992 * gst/rtsp-server/rtsp-stream.c:
10993 stream: set udp sources to PLAYING
10994 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10995 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10997 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10999 * gst/rtsp-server/rtsp-media-factory-uri.c:
11000 factory-uri: take ref to factory
11001 Take a ref to the factory that we place in our list.
11003 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11005 * tests/Makefile.am:
11006 * tests/test-reuse.c:
11007 test: add test for server reuse
11008 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
11010 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
11012 * gst/rtsp-server/rtsp-server.c:
11013 server: start and stop multiple times
11014 Stop listening on the RTSP port when the GSource is removed, so clients
11015 can't connect and the server can be started again.
11016 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
11018 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11020 * gst/rtsp-server/rtsp-server.c:
11021 server: fix small leak
11023 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11025 * gst/rtsp-server/rtsp-media.c:
11026 media: unref source in finish_unprepare
11027 The source is created in prepare, unref it in finish_unprepare.
11028 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
11030 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
11032 * gst/rtsp-server/rtsp-client.c:
11033 * gst/rtsp-server/rtsp-media.c:
11034 rtsp-media: remove bus watch before finalizing
11035 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
11036 * An extra media ref is added for the bus watch. This extra ref is unreffed by
11037 the GDestroyNotify function.
11038 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
11039 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
11040 gst_rtsp_media_unprepare before unreffing the media.
11041 This way, the bus watch will be removed before the media is finalized.
11042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
11044 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
11046 * gst/rtsp-server/rtsp-client.c:
11047 * gst/rtsp-server/rtsp-client.h:
11048 client: wait until the TEARDOWN response is sent to close the connection
11049 Responses can be sent async so we need to wait until the TEARDOWN response has
11050 been written before we close the connection to the client. This avoids the risk
11051 of writing/polling closed sockets.
11052 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
11054 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
11056 * gst/rtsp-server/rtsp-stream.c:
11057 rtsp-stream: plug socket leak
11058 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
11060 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
11063 Automatic update of common submodule
11064 From 6bb6951 to a72faea
11066 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
11068 * gst/rtsp-server/rtsp-media-factory-uri.c:
11069 rtsp-server: don't use deprecated API
11071 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
11073 * gst/rtsp-server/rtsp-client.c:
11074 rtsp-client: fix unused-but-set-variable compiler warning
11075 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
11077 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11080 * docs/libs/gst-rtsp-server-sections.txt:
11081 * gst/rtsp-server/rtsp-client.c:
11084 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11086 * examples/Makefile.am:
11087 * examples/test-multicast2.c:
11088 examples: add another multicast example
11089 Add an example for how to configure separate multicast ranges for each media
11092 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11094 * examples/test-multicast.c:
11097 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11099 * gst/rtsp-server/rtsp-client.c:
11100 * gst/rtsp-server/rtsp-media.c:
11101 * gst/rtsp-server/rtsp-session-media.c:
11102 * gst/rtsp-server/rtsp-session-media.h:
11103 * gst/rtsp-server/rtsp-stream-transport.c:
11104 * gst/rtsp-server/rtsp-stream-transport.h:
11105 stream: use the address managed by the stream
11106 Use the address managed by the stream for multicast. This allows us to have 1
11107 multicast address for each stream.
11108 Because the address is now managed by the stream we don't have to pass it around
11110 Set the address pool on the streams.
11112 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11114 * gst/rtsp-server/rtsp-client.c:
11115 * gst/rtsp-server/rtsp-media.c:
11116 * gst/rtsp-server/rtsp-stream.c:
11117 rtsp: improve debug
11119 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11121 * gst/rtsp-server/rtsp-media.c:
11122 * gst/rtsp-server/rtsp-media.h:
11123 media: add signal for new streams
11124 This allows applications to listen for new streams and configure properties on
11125 them, like the address pool.
11127 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11129 * gst/rtsp-server/rtsp-media.c:
11130 media: configure address pool in new streams
11132 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11134 * gst/rtsp-server/rtsp-stream.c:
11135 * gst/rtsp-server/rtsp-stream.h:
11136 stream: add methods to deal with address pool
11137 Add methods to get and set the address pool for the stream
11138 Add method to allocate and get the multicast addresses for this stream.
11140 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11142 * docs/libs/gst-rtsp-server-sections.txt:
11143 * gst/rtsp-server/rtsp-media.c:
11144 * gst/rtsp-server/rtsp-media.h:
11145 media: remove MTU property
11146 It is a stream property
11148 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11150 * gst/rtsp-server/rtsp-client.c:
11151 client: set blocksize only on stream
11152 Set the blocksize only on the current stream.
11154 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11156 * gst/rtsp-server/rtsp-stream.c:
11157 stream: share src and sink sockets
11158 the allocated socket is in the used-socket property, not socket.
11160 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11162 * gst/rtsp-server/rtsp-address-pool.c:
11163 * gst/rtsp-server/rtsp-address-pool.h:
11164 * gst/rtsp-server/rtsp-client.c:
11165 * gst/rtsp-server/rtsp-session-media.c:
11166 * gst/rtsp-server/rtsp-session-media.h:
11167 * gst/rtsp-server/rtsp-stream-transport.c:
11168 * gst/rtsp-server/rtsp-stream-transport.h:
11169 * tests/check/gst/addresspool.c:
11170 rtsp: make address-pool return an address object
11171 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
11172 store more info in the structure and allows us to more easily return the address
11173 to the right pool when no longer needed.
11174 Pass the address to the StreamTransport so that we can return it to the pool
11175 when the stream transport is freed or changed.
11177 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11179 * examples/Makefile.am:
11180 * examples/test-multicast.c:
11181 examples: add multicast example
11182 Show how to set up the multicast address pool so that media can be
11183 server with multicast.
11185 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11187 * gst/rtsp-server/rtsp-client.c:
11188 * gst/rtsp-server/rtsp-media-factory.c:
11189 * gst/rtsp-server/rtsp-media-factory.h:
11190 * gst/rtsp-server/rtsp-media.c:
11191 * gst/rtsp-server/rtsp-media.h:
11192 rtsp: use AddressPool
11193 Remove the multicast_group property.
11194 Use the configured addresspool to allocate multicast addresses.
11196 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11198 * gst/rtsp-server/rtsp-address-pool.c:
11199 * gst/rtsp-server/rtsp-address-pool.h:
11200 address-pool: add clear method
11202 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11204 * gst/rtsp-server/rtsp-address-pool.c:
11205 address-pool: small cleanups
11207 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11209 * tests/check/Makefile.am:
11210 * tests/check/gst/addresspool.c:
11211 tests: add addresspool unit test
11213 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11215 * gst/rtsp-server/Makefile.am:
11216 * gst/rtsp-server/rtsp-address-pool.c:
11217 * gst/rtsp-server/rtsp-address-pool.h:
11218 address-pool: add object to manage multicast addresses
11219 Make an object that can manage a rage of multicast addresses and ports.
11221 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11223 * gst/rtsp-server/rtsp-server.c:
11224 server: set default max-threads property
11226 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11228 * gst/rtsp-server/rtsp-media.c:
11229 media: wait for concurrent _prepare
11230 If a prepare is busy, wait for the result.
11232 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11234 * gst/rtsp-server/rtsp-media.c:
11235 media: add lock around message handler
11236 We don't want to dispatch messages while we are still processing the result of
11239 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11241 * gst/rtsp-server/rtsp-media.c:
11242 * gst/rtsp-server/rtsp-media.h:
11243 media: add lock to protect state changes
11245 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11247 * gst/rtsp-server/rtsp-stream.c:
11248 * gst/rtsp-server/rtsp-stream.h:
11249 stream: add locking
11251 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11253 * gst/rtsp-server/rtsp-stream-transport.c:
11254 * gst/rtsp-server/rtsp-stream-transport.h:
11255 * gst/rtsp-server/rtsp-stream.c:
11256 stream-transport: add keep-alive method
11258 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11260 * gst/rtsp-server/rtsp-stream-transport.c:
11261 * gst/rtsp-server/rtsp-stream-transport.h:
11262 * gst/rtsp-server/rtsp-stream.c:
11263 stream-transport: add method to handle RTP/RTCP
11264 Call new methods instead of poking into the structures directly.
11266 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11268 * gst/rtsp-server/rtsp-session-media.c:
11269 * gst/rtsp-server/rtsp-session-media.h:
11270 session-media: add locking
11272 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11274 * gst/rtsp-server/rtsp-session.c:
11275 * gst/rtsp-server/rtsp-session.h:
11276 session: add locking
11278 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11280 * gst/rtsp-server/rtsp-server.c:
11281 server: free old socket
11283 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11285 * gst/rtsp-server/rtsp-media-mapping.c:
11286 * gst/rtsp-server/rtsp-media-mapping.h:
11287 mapping: add locking
11289 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11291 * gst/rtsp-server/rtsp-media-factory.c:
11292 media-factory: add locking
11294 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11296 * gst/rtsp-server/rtsp-auth.c:
11297 * gst/rtsp-server/rtsp-auth.h:
11300 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11302 * gst/rtsp-server/rtsp-server.c:
11303 * gst/rtsp-server/rtsp-server.h:
11304 server: add max-thread property
11306 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11308 * gst/rtsp-server/rtsp-server.c:
11309 * gst/rtsp-server/rtsp-server.h:
11310 server: use a threadpool for the mainloops
11312 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11314 * gst/rtsp-server/rtsp-client.c:
11315 * gst/rtsp-server/rtsp-client.h:
11316 client: rename method
11317 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
11318 don't really create the client from the socket, we use the socket for the
11321 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11323 * gst/rtsp-server/rtsp-client.c:
11324 * gst/rtsp-server/rtsp-client.h:
11325 * gst/rtsp-server/rtsp-server.c:
11326 server: rework maincontext handling in clients
11327 Make a separate method to attach a client to a MainContext.
11328 Let the server decide in what GMainContext the client will operate and give this
11329 context to the client in attach. Then the server can later decide to use a
11330 separate thread for each client or just use the mainthread.
11332 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11334 * gst/rtsp-server/rtsp-client.c:
11335 * gst/rtsp-server/rtsp-session.c:
11336 * gst/rtsp-server/rtsp-session.h:
11337 session: move session header code in session object
11339 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
11343 * examples/test-auth.c:
11344 * examples/test-launch.c:
11345 * examples/test-mp4.c:
11346 * examples/test-ogg.c:
11347 * examples/test-readme.c:
11348 * examples/test-sdp.c:
11349 * examples/test-uri.c:
11350 * examples/test-video.c:
11351 * gst/rtsp-server/rtsp-auth.c:
11352 * gst/rtsp-server/rtsp-auth.h:
11353 * gst/rtsp-server/rtsp-client.c:
11354 * gst/rtsp-server/rtsp-client.h:
11355 * gst/rtsp-server/rtsp-media-factory-uri.c:
11356 * gst/rtsp-server/rtsp-media-factory-uri.h:
11357 * gst/rtsp-server/rtsp-media-factory.c:
11358 * gst/rtsp-server/rtsp-media-factory.h:
11359 * gst/rtsp-server/rtsp-media-mapping.c:
11360 * gst/rtsp-server/rtsp-media-mapping.h:
11361 * gst/rtsp-server/rtsp-media.c:
11362 * gst/rtsp-server/rtsp-media.h:
11363 * gst/rtsp-server/rtsp-params.c:
11364 * gst/rtsp-server/rtsp-params.h:
11365 * gst/rtsp-server/rtsp-sdp.c:
11366 * gst/rtsp-server/rtsp-sdp.h:
11367 * gst/rtsp-server/rtsp-server.c:
11368 * gst/rtsp-server/rtsp-server.h:
11369 * gst/rtsp-server/rtsp-session-media.c:
11370 * gst/rtsp-server/rtsp-session-media.h:
11371 * gst/rtsp-server/rtsp-session-pool.c:
11372 * gst/rtsp-server/rtsp-session-pool.h:
11373 * gst/rtsp-server/rtsp-session.c:
11374 * gst/rtsp-server/rtsp-session.h:
11375 * gst/rtsp-server/rtsp-stream-transport.c:
11376 * gst/rtsp-server/rtsp-stream-transport.h:
11377 * gst/rtsp-server/rtsp-stream.c:
11378 * gst/rtsp-server/rtsp-stream.h:
11379 * tests/check/gst/rtspserver.c:
11380 * tests/test-cleanup.c:
11383 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11385 * gst/rtsp-server/rtsp-media.c:
11386 * gst/rtsp-server/rtsp-session-media.c:
11387 * gst/rtsp-server/rtsp-session.c:
11388 rtsp-server: added annotations to indicate type of ownership transfer of return values
11389 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11391 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11394 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11396 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11399 * bindings/Makefile.am:
11400 * bindings/vala/Makefile.am:
11401 * bindings/vala/gst-rtsp-server-0.10.deps:
11402 * bindings/vala/gst-rtsp-server-0.10.vapi:
11403 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11404 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11405 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11406 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11407 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11409 bindings: remove vala bindings
11410 They'll be reunited with the other GStreamer bindings
11411 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11413 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11415 * gst/rtsp-server/rtsp-client.c:
11416 * gst/rtsp-server/rtsp-session-media.c:
11417 * gst/rtsp-server/rtsp-session-media.h:
11418 * gst/rtsp-server/rtsp-stream-transport.c:
11419 * gst/rtsp-server/rtsp-stream-transport.h:
11420 rtsp: only create transport when needed
11421 Only create the StreamTransport when configured.
11423 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11425 * gst/rtsp-server/rtsp-client.c:
11426 client: small cleanup
11428 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11430 * gst/rtsp-server/rtsp-client.c:
11431 * gst/rtsp-server/rtsp-client.h:
11432 * gst/rtsp-server/rtsp-stream-transport.c:
11433 * gst/rtsp-server/rtsp-stream-transport.h:
11434 rtsp: refactor configuration of transport
11435 Move the configuration of the transport to a place where it makes
11438 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11440 * gst/rtsp-server/rtsp-client.c:
11441 client: refactor transport parsing
11443 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11445 * gst/rtsp-server/rtsp-client.c:
11446 client: refuse to change the MTU on shared media
11447 If we change the MTU of chared media, it changes for all clients.
11448 We don't want to set the MTU to something large for clients that
11451 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11453 * examples/test-mp4.c:
11454 * gst/rtsp-server/rtsp-media.c:
11455 small fixes to docs and debug
11457 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11459 * gst/rtsp-server/rtsp-stream.c:
11460 stream: transports must already have been removed
11462 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11464 * gst/rtsp-server/rtsp-media.c:
11465 * gst/rtsp-server/rtsp-stream.c:
11466 * gst/rtsp-server/rtsp-stream.h:
11467 stream: improve join and leave of the pipeline
11469 Do the cleanup properly
11472 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11474 * gst/rtsp-server/rtsp-media.c:
11475 media: move unprepare below default implementation
11476 Makes it easier to find the default implementation
11478 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11480 * gst/rtsp-server/rtsp-media.c:
11481 media: signal unprepared when we actually finish
11483 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11485 * gst/rtsp-server/rtsp-media.c:
11486 media: no need to unlock, unprepare does that when needed
11488 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11490 * docs/libs/gst-rtsp-server-sections.txt:
11491 * gst/rtsp-server/rtsp-media-factory.h:
11492 * gst/rtsp-server/rtsp-media-mapping.c:
11493 * gst/rtsp-server/rtsp-media.h:
11494 * gst/rtsp-server/rtsp-params.c:
11495 * gst/rtsp-server/rtsp-server.c:
11496 * gst/rtsp-server/rtsp-session-pool.h:
11497 * gst/rtsp-server/rtsp-session.c:
11498 * gst/rtsp-server/rtsp-session.h:
11499 * gst/rtsp-server/rtsp-stream-transport.h:
11500 * gst/rtsp-server/rtsp-stream.h:
11503 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11505 * gst/rtsp-server/rtsp-client.c:
11506 * gst/rtsp-server/rtsp-media-mapping.h:
11507 * gst/rtsp-server/rtsp-media.c:
11508 * gst/rtsp-server/rtsp-media.h:
11509 * gst/rtsp-server/rtsp-server.h:
11510 * gst/rtsp-server/rtsp-stream.c:
11511 * gst/rtsp-server/rtsp-stream.h:
11512 rtsp: fix MTU setting
11513 Fix setting of the MTU. There is no need for a vmethod.
11515 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11520 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11523 configure: bump version number after refactoring
11525 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11527 * gst/rtsp-server/Makefile.am:
11528 * gst/rtsp-server/rtsp-client.c:
11529 * gst/rtsp-server/rtsp-client.h:
11530 * gst/rtsp-server/rtsp-media-factory-uri.c:
11531 * gst/rtsp-server/rtsp-media-factory.c:
11532 * gst/rtsp-server/rtsp-media-factory.h:
11533 * gst/rtsp-server/rtsp-media.c:
11534 * gst/rtsp-server/rtsp-media.h:
11535 * gst/rtsp-server/rtsp-sdp.c:
11536 * gst/rtsp-server/rtsp-session-media.c:
11537 * gst/rtsp-server/rtsp-session-media.h:
11538 * gst/rtsp-server/rtsp-session.c:
11539 * gst/rtsp-server/rtsp-session.h:
11540 * gst/rtsp-server/rtsp-stream-transport.c:
11541 * gst/rtsp-server/rtsp-stream-transport.h:
11542 * gst/rtsp-server/rtsp-stream.c:
11543 * gst/rtsp-server/rtsp-stream.h:
11544 rtsp: massive refactoring
11545 Make GObjects from the remaining simple structures.
11546 Remove GstRTSPSessionStream, it's not needed.
11547 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11548 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11549 a GstRTSPStream should be transported to a client.
11550 Rename GstRTSPMediaFactory::get_element -> create_element because that
11551 more accurately describes what it does.
11552 Make nice methods instead of poking in the structures.
11553 Move some methods inside the relevant object source code.
11554 Use GPtrArray to store objects instead of plain arrays, it is more
11555 natural and allows us to more easily clean up.
11556 Move the allocation of udp ports to the Stream object. The Stream object
11557 contains the elements needed to stream the media to a client.
11558 Improve the prepare and unprepare methods. Unprepare should now undo
11559 everything prepare did. Improve also async unprepare when doing EOS on
11560 shutdown. Make sure we always unprepare correctly.
11562 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11564 * gst/rtsp-server/rtsp-client.c:
11565 rtsp-client: Unref server address clients connected to
11566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11568 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11570 * gst/rtsp-server/rtsp-server.c:
11571 rtsp-server: don't ref server socket if it is NULL
11572 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11573 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11575 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11577 * tests/check/Makefile.am:
11578 tests: Add libgio link dependency
11579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11581 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11583 * gst/rtsp-server/rtsp-media-mapping.c:
11584 * gst/rtsp-server/rtsp-media-mapping.h:
11585 rtsp-media-mapping: rename find_media vfunc to find_factory
11586 The virtual method and class method should have the same name
11587 so it is correctly represented in GIR file
11588 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11590 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11592 * gst/rtsp-server/rtsp-auth.c:
11593 * gst/rtsp-server/rtsp-client.c:
11594 * gst/rtsp-server/rtsp-media-factory-uri.c:
11595 * gst/rtsp-server/rtsp-media-factory.c:
11596 * gst/rtsp-server/rtsp-media-mapping.c:
11597 * gst/rtsp-server/rtsp-media.c:
11598 * gst/rtsp-server/rtsp-server.c:
11599 * gst/rtsp-server/rtsp-session-pool.c:
11600 * gst/rtsp-server/rtsp-session.c:
11601 rtsp-server: fixed comments and GIR annotations
11602 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11604 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11606 * gst/rtsp-server/rtsp-media-mapping.c:
11607 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11609 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11611 * gst/rtsp-server/rtsp-server.c:
11612 rtsp-server: allow binding on port 0 (binds on a random port)
11614 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11616 * gst/rtsp-server/rtsp-server.c:
11617 * gst/rtsp-server/rtsp-server.h:
11618 rtsp-server: add bound-port property
11619 bound-port can be used to retrieve the port number when the server is bound on
11620 port 0, which binds on a random port.
11622 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11624 * gst/rtsp-server/rtsp-media-factory.c:
11625 * gst/rtsp-server/rtsp-media-factory.h:
11626 rtsp-media-factory: make ::get_element overridable by GI bindings
11627 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11628 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11629 as the invoker for ::get_element(), making it overridable by GI generated
11632 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11634 * gst/rtsp-server/rtsp-media-factory-uri.c:
11635 rtsp-media-factory-uri: don't autoplug parsers in a loop
11636 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11639 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11641 * gst/rtsp-server/Makefile.am:
11642 Explicitly link against gio. Fix link error on mac.
11644 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11646 * gst/rtsp-server/rtsp-session.c:
11647 session: add ttl to the transport header in SETUP
11648 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11650 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11652 * gst/rtsp-server/rtsp-client.c:
11653 * gst/rtsp-server/rtsp-client.h:
11654 * gst/rtsp-server/rtsp-media.c:
11655 client: Use client transport settings for multicast if allowed.
11656 This patch makes it possible for the client to send transport settings for
11657 multicast (destination && ttl). Client settings must be explicitly allowed or
11658 the server will use its own settings.
11659 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11661 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11664 Automatic update of common submodule
11665 From 6c0b52c to 6bb6951
11667 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11669 * gst/rtsp-server/rtsp-client.c:
11670 rtsp-client: do not destroy the rtsp watch
11671 Don't destroy the client watch while dispatching. The rtsp watch is
11672 automatically destroyed after the rtsp watch function closed() has
11674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11676 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11679 Automatic update of common submodule
11680 From 4f962f7 to 6c0b52c
11682 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11684 * gst/rtsp-server/rtsp-media.c:
11685 media: fix check for seekability
11687 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11689 * gst/rtsp-server/rtsp-client.c:
11690 client: use more GIO
11691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11693 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11695 * gst/rtsp-server/rtsp-server.c:
11696 server: remove obsolete includes
11698 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11700 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11701 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11702 be available in "on_new_ssrc". The transports are added in
11703 gst_rtsp_media_set_state when going to PLAYING state. However,
11704 "on_new_ssrc" might be called before this happens.
11705 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11707 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11709 * gst/rtsp-server/rtsp-client.c:
11710 * gst/rtsp-server/rtsp-client.h:
11711 rtsp-client: add signals for rtsp requests (fixes #683287)
11713 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11715 * gst/rtsp-server/rtsp-client.c:
11716 * gst/rtsp-server/rtsp-client.h:
11717 add new-session signal to rtsp-client (fixes #683058)
11719 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11722 Automatic update of common submodule
11723 From 668acee to 4f962f7
11725 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11727 * gst/rtsp-server/rtsp-server.c:
11728 * tests/check/gst/rtspserver.c:
11729 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11730 Do not assume that *error is set in g_socket_address_enumerator_next.
11731 Added test_bind_already_in_use unit-test.
11732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11734 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11737 Automatic update of common submodule
11738 From 94ccf4c to 668acee
11740 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11742 * gst/rtsp-server/rtsp-client.c:
11743 * gst/rtsp-server/rtsp-client.h:
11744 rtsp-client: make create_sdp virtual method
11745 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11747 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11750 Automatic update of common submodule
11751 From 98e386f to 94ccf4c
11753 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11755 * gst/rtsp-server/rtsp-client.c:
11758 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11760 * gst/rtsp-server/rtsp-client.c:
11761 * gst/rtsp-server/rtsp-client.h:
11762 * gst/rtsp-server/rtsp-server.c:
11763 * gst/rtsp-server/rtsp-server.h:
11764 rtsp-server: use an existing socket to establish HTTP tunnel
11765 Make it possible to transfer a socket from an HTTP server to be used as
11766 an RTSP over HTTP tunnel.
11768 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11770 * gst/rtsp-server/rtsp-client.c:
11771 * gst/rtsp-server/rtsp-media.c:
11772 * gst/rtsp-server/rtsp-media.h:
11773 rtsp: Handle the blocksize parameter
11774 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11776 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11778 * tests/check/Makefile.am:
11779 * tests/check/gst/rtspserver.c:
11780 Have unit test get header from source dir, not installed dir
11781 This makes compilation of unit tests work in a build directory other
11782 than the source directory.
11783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11785 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11787 * gst/rtsp-server/rtsp-media.c:
11788 rtsp-media: update for gst_element_make_from_uri() changes
11790 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11793 * tests/Makefile.am:
11794 * tests/check/Makefile.am:
11795 * tests/check/gst/rtspserver.c:
11796 rtsp: add unit test
11797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11799 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11801 * gst/rtsp-server/rtsp-media.c:
11802 rtsp-media: don't collect media stats when going to NULL
11803 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11805 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11807 * gst/rtsp-server/rtsp-client.c:
11808 client: don't leak transports
11810 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11812 * gst/rtsp-server/rtsp-client.c:
11813 rtsp-client: free transport on no_stream in SETUP handler
11815 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11817 * gst/rtsp-server/rtsp-client.c:
11818 rtsp-client: changed session media iteration
11819 In client_unlink_session: now don't iterate in session->medias
11820 list where items are removed by gst_rtsp_session_release_media.
11821 Instead, repeatedly remove the first item.
11823 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11825 * gst/rtsp-server/rtsp-client.c:
11826 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11827 GstRTSPSessionMedia is not a GObject type. When the
11828 GstRTSPSession is freed, it will free the media.
11830 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11832 * gst/rtsp-server/rtsp-media-factory.c:
11833 factory: plug pad leak in collect_streams
11834 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11835 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11836 will take one reference, and the other reference will otherwise
11837 give a memory leak.
11839 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11842 configure: suppress some warnings when debug is disabled
11843 Warnings about unused variables should be suppressed if core has the
11844 debug system disabled.
11845 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11847 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11849 * docs/libs/Makefile.am:
11850 docs: fix build in uninstalled setup
11851 Include gst-plugins-base libs properly.
11853 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11855 * docs/libs/gst-rtsp-server.types:
11856 docs: include headers defining rtsp-server object types
11857 Fixes compiler warnings during docs build.
11858 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11860 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11863 configure: Add warning flags for compiler when configuring
11864 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11866 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11869 Automatic update of common submodule
11870 From 03a0e57 to 98e386f
11872 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11875 Automatic update of common submodule
11876 From 1fab359 to 03a0e57
11878 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11880 * gst/rtsp-server/rtsp-client.c:
11881 client: fix GSocketAddress leak in gst_rtsp_client_accept
11882 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11884 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11887 Automatic update of common submodule
11888 From f1b5a96 to 1fab359
11890 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11893 Automatic update of common submodule
11894 From 92b7266 to f1b5a96
11896 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11899 Automatic update of common submodule
11900 From ec1c4a8 to 92b7266
11902 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11905 Automatic update of common submodule
11906 From 3429ba6 to ec1c4a8
11908 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11910 * gst/rtsp-server/rtsp-auth.c:
11911 * gst/rtsp-server/rtsp-client.c:
11912 * gst/rtsp-server/rtsp-media-factory-uri.c:
11913 * gst/rtsp-server/rtsp-server.c:
11914 rtsp: fix compiler warnings
11915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11917 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11920 Automatic update of common submodule
11921 From dc70203 to 3429ba6
11923 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11925 * gst/rtsp-server/rtsp-client.c:
11926 * gst/rtsp-server/rtsp-media-factory.c:
11927 * gst/rtsp-server/rtsp-media-factory.h:
11928 * gst/rtsp-server/rtsp-media.c:
11929 * gst/rtsp-server/rtsp-media.h:
11930 * gst/rtsp-server/rtsp-server.c:
11931 * gst/rtsp-server/rtsp-server.h:
11932 * gst/rtsp-server/rtsp-session-pool.c:
11933 * gst/rtsp-server/rtsp-session-pool.h:
11934 rtsp-server: port to new thread API
11936 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11939 Automatic update of common submodule
11940 From 6db25be to dc70203
11942 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11944 * gst/rtsp-server/rtsp-auth.c:
11945 * gst/rtsp-server/rtsp-auth.h:
11946 * gst/rtsp-server/rtsp-client.c:
11947 rtsp-server: Fix compilation and compiler warnings
11949 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11953 * gst/rtsp-server/Makefile.am:
11954 configure: Modernize autotools setup a bit
11955 Also we now only create tar.bz2 and tar.xz tarballs.
11957 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11960 Automatic update of common submodule
11961 From 464fe15 to 6db25be
11963 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11966 Automatic update of common submodule
11967 From 7fda524 to 464fe15
11969 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11972 * docs/libs/Makefile.am:
11973 * docs/version.entities.in:
11974 * gst-rtsp.spec.in:
11975 * gst/rtsp-server/Makefile.am:
11976 * pkgconfig/Makefile.am:
11977 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11978 * pkgconfig/gstreamer-rtsp-server.pc.in:
11979 * tests/Makefile.am:
11980 rtsp-server: Update versioning
11982 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11984 Merge remote-tracking branch 'origin/0.10'
11986 gst/rtsp-server/rtsp-session-pool.c
11988 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11990 * gst/rtsp-server/rtsp-session-pool.c:
11991 rtsp-server: Don't use deprecated GLib API
11993 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11995 Replace master with 0.11
11997 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11999 Merge branch 'master' into 0.11
12001 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12003 Merge branch 'master' into 0.11
12005 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
12008 A couple minor typo fixes
12010 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12012 * gst/rtsp-server/rtsp-media.c:
12013 media: fix state of the appqueue
12015 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12017 * gst/rtsp-server/rtsp-media-factory-uri.c:
12018 factory: use videoconvert
12020 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12022 * gst/rtsp-server/rtsp-media-factory-uri.c:
12023 factory: change to new style caps
12025 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12027 * gst/rtsp-server/rtsp-client.c:
12028 * gst/rtsp-server/rtsp-client.h:
12029 * gst/rtsp-server/rtsp-media-factory-uri.c:
12030 * gst/rtsp-server/rtsp-media.c:
12031 * gst/rtsp-server/rtsp-server.c:
12032 * gst/rtsp-server/rtsp-server.h:
12033 * gst/rtsp-server/rtsp-session-pool.c:
12034 rtsp-server: port to GIO
12037 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12040 configure: fix build
12042 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12045 docs: fix for gst_rtsp_server_set_port() -> _set_service()
12046 https://bugzilla.gnome.org/show_bug.cgi?id=666548
12048 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12051 * examples/Makefile.am:
12052 First rule of gst-rtsp-server club: don't talk about gst-phonon
12054 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12057 * pkgconfig/Makefile.am:
12058 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
12059 * pkgconfig/gstreamer-rtsp-server.pc.in:
12060 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
12061 For consistency with all other modules.
12063 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12065 * gst/rtsp-server/rtsp-client.c:
12066 rtsp-client: update for new map API
12068 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12071 * bindings/Makefile.am:
12072 * bindings/python/Makefile.am:
12073 * bindings/python/arg-types.py:
12074 * bindings/python/codegen/Makefile.am:
12075 * bindings/python/codegen/__init__.py:
12076 * bindings/python/codegen/argtypes.py:
12077 * bindings/python/codegen/code-coverage.py:
12078 * bindings/python/codegen/codegen.py:
12079 * bindings/python/codegen/definitions.py:
12080 * bindings/python/codegen/defsparser.py:
12081 * bindings/python/codegen/docextract.py:
12082 * bindings/python/codegen/docgen.py:
12083 * bindings/python/codegen/fileprefix.override:
12084 * bindings/python/codegen/fileprefixmodule.c:
12085 * bindings/python/codegen/h2def.py:
12086 * bindings/python/codegen/mergedefs.py:
12087 * bindings/python/codegen/mkskel.py:
12088 * bindings/python/codegen/override.py:
12089 * bindings/python/codegen/reversewrapper.py:
12090 * bindings/python/codegen/scmexpr.py:
12091 * bindings/python/rtspserver-types.defs:
12092 * bindings/python/rtspserver.defs:
12093 * bindings/python/rtspserver.override:
12094 * bindings/python/rtspservermodule.c:
12095 * bindings/python/test.py:
12097 python: remove pygst-based python bindings
12098 pygi is the future, apparently.
12100 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
12103 Automatic update of common submodule
12104 From c463bc0 to 7fda524
12106 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12109 Automatic update of common submodule
12110 From 2a59016 to c463bc0
12112 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12115 Automatic update of common submodule
12116 From 0807187 to 2a59016
12118 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12121 Automatic update of common submodule
12122 From 11f0cd5 to 0807187
12124 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12126 * examples/test-auth.c:
12127 example: update for new caps
12129 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12131 * examples/test-video.c:
12132 * gst/rtsp-server/rtsp-client.c:
12133 * gst/rtsp-server/rtsp-media-factory-uri.c:
12134 * gst/rtsp-server/rtsp-media.c:
12135 * gst/rtsp-server/rtsp-media.h:
12136 * gst/rtsp-server/rtsp-session.c:
12137 * gst/rtsp-server/rtsp-session.h:
12138 rtsp-server: port some more to 0.11
12140 Remove bufferlist stuff
12141 Update for new API.
12142 Add queue before appsink now that preroll-queue-len is gone.
12143 Update for request pad changes.
12145 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12147 Merge branch 'master' into 0.11
12149 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12151 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12152 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12153 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12155 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12157 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12158 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12159 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12161 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12163 Merge branch 'master' into 0.11
12165 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12167 * gst/rtsp-server/rtsp-media.c:
12168 * gst/rtsp-server/rtsp-media.h:
12169 media: add a seekable boolean
12170 Maintain the seekable state with a new variable instead of reusing the
12173 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
12175 * gst/rtsp-server/rtsp-media.c:
12176 Disallow seek in live media
12178 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12180 Merge branch 'master' into 0.11
12182 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
12184 * gst/rtsp-server/rtsp-server.c:
12185 #ifdef statements for windows socket creation were missing
12187 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
12190 Automatic update of common submodule
12191 From a39eb83 to 11f0cd5
12193 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
12196 Automatic update of common submodule
12197 From 605cd9a to a39eb83
12199 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12201 Merge branch 'master' into 0.11
12203 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12205 * gst/rtsp-server/rtsp-client.c:
12206 client: use method to access property
12208 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12210 * gst/rtsp-server/rtsp-media-factory.c:
12211 * gst/rtsp-server/rtsp-media-factory.h:
12212 media-factory: add protocols property
12213 Add a property to configure the allowed protocols in the media created from the
12216 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12218 * gst/rtsp-server/rtsp-media-factory.c:
12219 * gst/rtsp-server/rtsp-media-factory.h:
12220 media-factory: add media-configure signal
12221 Add signal to allow the application to configure the media after it was created
12224 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12226 * gst/rtsp-server/rtsp-client.c:
12227 client: use method to access property
12229 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12231 * gst/rtsp-server/rtsp-media-factory.c:
12232 * gst/rtsp-server/rtsp-media-factory.h:
12233 media-factory: add protocols property
12234 Add a property to configure the allowed protocols in the media created from the
12237 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12239 * gst/rtsp-server/rtsp-media-factory.c:
12240 * gst/rtsp-server/rtsp-media-factory.h:
12241 media-factory: add media-configure signal
12242 Add signal to allow the application to configure the media after it was created
12245 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12247 Merge branch 'master' into 0.11
12249 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12251 * gst/rtsp-server/rtsp-client.c:
12252 client: use media multicast group
12254 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12256 * gst/rtsp-server/rtsp-media-factory.h:
12257 * gst/rtsp-server/rtsp-server.h:
12258 * gst/rtsp-server/rtsp-session-pool.h:
12259 * gst/rtsp-server/rtsp-session.h:
12262 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12264 * gst/rtsp-server/rtsp-client.c:
12265 * gst/rtsp-server/rtsp-sdp.h:
12266 sdp: copy and free the server ip address
12267 Copy and free the server ip address to make memory management easier later.
12269 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12271 * gst/rtsp-server/rtsp-media-factory.c:
12272 media-factory: configure multicast in media
12274 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12276 * gst/rtsp-server/rtsp-media.c:
12277 * gst/rtsp-server/rtsp-media.h:
12278 media: add property for multicast group
12279 Add a property to configure the multicast group in the media.
12280 Based on patches from Marc Leeman and Robert Krakora.
12282 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12284 * gst/rtsp-server/rtsp-media-factory.c:
12285 * gst/rtsp-server/rtsp-media-factory.h:
12286 media-factory: add property for multicast group
12287 Add a property to configure the multicast group in the media factory.
12288 Based on patches from Marc Leeman and Robert Krakora.
12290 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12292 * gst/rtsp-server/rtsp-client.c:
12293 client: do configuration of transport in one place
12294 Move the configuration of the transport destination address to where we also
12295 configure the other bits.
12297 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12299 * gst/rtsp-server/rtsp-client.c:
12300 client: use media multicast group
12302 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12304 * gst/rtsp-server/rtsp-media-factory.h:
12305 * gst/rtsp-server/rtsp-server.h:
12306 * gst/rtsp-server/rtsp-session-pool.h:
12307 * gst/rtsp-server/rtsp-session.h:
12310 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12312 * gst/rtsp-server/rtsp-client.c:
12313 * gst/rtsp-server/rtsp-sdp.h:
12314 sdp: copy and free the server ip address
12315 Copy and free the server ip address to make memory management easier later.
12317 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12319 * gst/rtsp-server/rtsp-media-factory.c:
12320 media-factory: configure multicast in media
12322 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12324 * gst/rtsp-server/rtsp-media.c:
12325 * gst/rtsp-server/rtsp-media.h:
12326 media: add property for multicast group
12327 Add a property to configure the multicast group in the media.
12328 Based on patches from Marc Leeman and Robert Krakora.
12330 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12332 * gst/rtsp-server/rtsp-media-factory.c:
12333 * gst/rtsp-server/rtsp-media-factory.h:
12334 media-factory: add property for multicast group
12335 Add a property to configure the multicast group in the media factory.
12336 Based on patches from Marc Leeman and Robert Krakora.
12338 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12340 * gst/rtsp-server/rtsp-client.c:
12341 client: do configuration of transport in one place
12342 Move the configuration of the transport destination address to where we also
12343 configure the other bits.
12345 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12347 Merge branch 'master' into 0.11
12349 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12351 * gst/rtsp-server/rtsp-client.c:
12352 client: destroy pipeline on client disconnect with no prior TEARDOWN.
12353 The problem occurs when the client abruptly closes the connection without
12354 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
12355 server is where the pipeline gets torn down. Since this handler is not called,
12356 the pipeline remains and is up and running. Subsequent clients get their own
12357 pipelines and if the do not issue TEARDOWNs then those pipelines will also
12358 remain up and running. This is a resource leak.
12360 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12362 Merge branch 'master' into 0.11
12364 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
12366 * gst/rtsp-server/rtsp-media-factory.c:
12367 * gst/rtsp-server/rtsp-media-factory.h:
12368 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
12369 For example, it can be used to retrieve source elements like appsrc, in a more
12370 convenient way than subclassing get_element.
12372 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12374 Merge branch 'master' into 0.11
12376 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
12378 * gst/rtsp-server/rtsp-server.c:
12379 rtsp-server: hold on to reference while using object
12381 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12383 * gst/rtsp-server/rtsp-media.c:
12386 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12389 configure: use unstable api
12391 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12393 * gst/rtsp-server/rtsp-client.c:
12394 client: fix reference counting
12396 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12398 * gst/rtsp-server/rtsp-client.c:
12399 * gst/rtsp-server/rtsp-media.c:
12400 fix compiler warnings about unused variables
12402 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12404 * examples/test-launch.c:
12405 * examples/test-readme.c:
12406 * examples/test-uri.c:
12407 * examples/test-video.c:
12408 examples: tell rtsp uri when ready
12410 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12413 Automatic update of common submodule
12414 From 69b981f to 605cd9a
12416 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12418 * gst/rtsp-server/rtsp-client.c:
12419 client: update for buffer API change
12421 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12423 * gst/rtsp-server/Makefile.am:
12424 Makefile.am: 0.10 => @GST_MAJORMINOR@
12426 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12428 * gst/rtsp-server/rtsp-media-factory-uri.c:
12429 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12431 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12433 * gst/rtsp-server/.gitignore:
12434 .gitignore: 0.10 => 0.11
12436 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12438 * gst/rtsp-server/Makefile.am:
12439 Makefile.am: 0.10 => @GST_MAJORMINOR@
12441 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12443 Merge branch 'master' into 0.11
12445 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12448 Automatic update of common submodule
12449 From 9e5bbd5 to 69b981f
12451 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12454 Automatic update of common submodule
12455 From fd35073 to 9e5bbd5
12457 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12460 Automatic update of common submodule
12461 From 46dfcea to fd35073
12463 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12465 * gst/rtsp-server/rtsp-media-factory-uri.c:
12466 * gst/rtsp-server/rtsp-media.c:
12467 media: port to new caps API
12469 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12471 Merge branch 'master' into 0.11
12473 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12475 * bindings/vala/gst-rtsp-server-0.10.vapi:
12476 Updated Vala bindings.
12477 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12479 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12481 * gst/rtsp-server/rtsp-server.c:
12482 * gst/rtsp-server/rtsp-server.h:
12483 Add a signal for newly connected clients.
12484 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12486 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12488 * bindings/python/rtspserver.override:
12489 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12491 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12493 * gst/rtsp-server/Makefile.am:
12494 * gst/rtsp-server/rtsp-client.c:
12495 * gst/rtsp-server/rtsp-funnel.c:
12496 * gst/rtsp-server/rtsp-funnel.h:
12497 * gst/rtsp-server/rtsp-media.c:
12498 rtsp-server: port to 0.11
12500 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12505 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12507 Merge branch 'master' into 0.11
12512 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12515 Automatic update of common submodule
12516 From c3cafe1 to 46dfcea
12518 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12520 * bindings/python/Makefile.am:
12521 * bindings/python/rtspserver.defs:
12522 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12524 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12526 * bindings/python/arg-types.py:
12527 python bindings: add GstRTSPUrlParam
12528 Needed to implement MediaFactory virtual proxies
12530 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12532 * bindings/python/arg-types.py:
12533 python bindings: fix returning GstRTSPUrl types
12535 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12537 * bindings/python/arg-types.py:
12538 python bindings: add arg type for GstRTSPUrl
12540 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12542 * bindings/python/rtspserver.defs:
12543 python bindings: fix the definition of MediaFactory.collect_stream
12545 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12548 Automatic update of common submodule
12549 From 1ccbe09 to c3cafe1
12551 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12554 Automatic update of common submodule
12555 From 193b717 to 1ccbe09
12557 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12560 Automatic update of common submodule
12561 From b77e2bf to 193b717
12563 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12566 build: Include lcov.mak to allow test coverage report generation
12568 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12571 Automatic update of common submodule
12572 From d8814b6 to b77e2bf
12574 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12577 Automatic update of common submodule
12578 From 6aaa286 to d8814b6
12580 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12583 Automatic update of common submodule
12584 From 6aec6b9 to 6aaa286
12586 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12589 autogen: wingo signed comment
12591 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12593 * gst/rtsp-server/rtsp-session-pool.c:
12594 session: use full charset for RTSP session ID
12595 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12596 session ID more difficult.
12597 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12599 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12601 * gst/rtsp-server/Makefile.am:
12602 rtsp-server: Don't install the funnel header
12604 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12607 Automatic update of common submodule
12608 From 1de7f6a to 6aec6b9
12610 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12613 configure: require core/base 0.10.31
12614 Needed at least for gst_plugin_feature_rank_compare_func().
12616 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12619 Automatic update of common submodule
12620 From f94d739 to 1de7f6a
12622 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12624 * gst/rtsp-server/rtsp-media.c:
12625 media: remove more unused code
12627 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12629 * gst/rtsp-server/rtsp-media.c:
12630 * gst/rtsp-server/rtsp-media.h:
12631 media: remove duplicate filtering
12632 Remove the duplicate filtering code now that we have a released -good version.
12633 Give a warning instead.
12635 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12637 * gst/rtsp-server/rtsp-media-factory.c:
12638 * gst/rtsp-server/rtsp-media.c:
12639 media: fix default buffer size
12641 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12643 * gst/rtsp-server/rtsp-media-factory.c:
12644 * gst/rtsp-server/rtsp-media-factory.h:
12645 media-factory: add property to configure the buffer-size
12646 Add a property to configure the kernel UDP buffer size.
12648 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12650 * gst/rtsp-server/rtsp-media.c:
12651 * gst/rtsp-server/rtsp-media.h:
12652 media: add property to configure kernel buffer sizes
12653 Add a property to configure the kernel UDP buffer size.
12655 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12658 configure: set PYGOBJECT_REQ before using it
12659 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12661 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12663 * docs/Makefile.am:
12664 docs: recursive into sub-directories on 'make upload'
12666 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12668 * docs/libs/gst-rtsp-server-docs.sgml:
12669 * docs/version.entities.in:
12670 docs: mention full version these docs are for, not just major-minor
12672 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12675 back to development
12677 === release 0.10.8 ===
12679 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12684 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12686 * gst/rtsp-server/rtsp-server.c:
12687 rtsp-server: clarify docs a little
12689 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12691 * gst/rtsp-server/rtsp-media.c:
12692 media: init debug category before starting thread
12694 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12696 * gst/rtsp-server/rtsp-auth.c:
12697 auth: add realm to make it more spec compliant
12699 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12701 * gst/rtsp-server/rtsp-server.c:
12702 * gst/rtsp-server/rtsp-server.h:
12703 server: add locking
12705 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12707 * examples/test-video.c:
12708 example: improve example docs a little
12710 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12712 * gst/rtsp-server/rtsp-server.c:
12713 server: ensure the watch has a ref to the server
12715 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12717 * gst/rtsp-server/rtsp-server.c:
12718 server: simpify channel function
12720 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12722 * gst/rtsp-server/rtsp-server.c:
12723 * gst/rtsp-server/rtsp-server.h:
12724 server: simplify management of channel and source
12725 We don't need to keep around the channel and source objects. Let the mainloop
12726 and the source manage the source and channel respectively.
12728 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12734 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12736 * tests/.gitignore:
12737 * tests/Makefile.am:
12738 * tests/test-cleanup.c:
12739 tests: add tests directory and cleanup test
12741 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12743 * gst/rtsp-server/rtsp-media-factory-uri.c:
12744 * gst/rtsp-server/rtsp-media-factory.c:
12745 * gst/rtsp-server/rtsp-media-mapping.c:
12746 * gst/rtsp-server/rtsp-media.c:
12747 * gst/rtsp-server/rtsp-session-pool.c:
12748 * gst/rtsp-server/rtsp-session.c:
12749 server: improve debugging in various objects
12751 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12753 * gst/rtsp-server/rtsp-server.c:
12754 server: chain up to the parent finalize
12756 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12758 * bindings/python/rtspserver-types.defs:
12759 * bindings/python/rtspserver.defs:
12760 * bindings/python/rtspserver.override:
12761 * bindings/python/test.py:
12762 gst-rtsp-server: update python bindings
12764 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12766 * gst/rtsp-server/rtsp-client.c:
12767 client: use the response from the clientstate
12768 Create the response object only once and store in the client state.
12769 Make all methods use the state response,
12771 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12773 * gst/rtsp-server/rtsp-server.c:
12774 server: use signal to keep track of clients
12775 Keep track of all the clients that the server creates and remove them when they
12776 fire the 'closed' signal.
12778 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12780 * gst/rtsp-server/rtsp-client.c:
12781 * gst/rtsp-server/rtsp-client.h:
12782 client: emit signal when closing
12784 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12786 * examples/.gitignore:
12787 * examples/Makefile.am:
12788 * examples/test-auth.c:
12789 * examples/test-video.c:
12790 * gst/rtsp-server/rtsp-auth.c:
12791 * gst/rtsp-server/rtsp-auth.h:
12792 * gst/rtsp-server/rtsp-client.c:
12793 * gst/rtsp-server/rtsp-media-factory.c:
12794 * gst/rtsp-server/rtsp-media.c:
12795 * gst/rtsp-server/rtsp-media.h:
12796 * gst/rtsp-server/rtsp-session-pool.h:
12797 * gst/rtsp-server/rtsp-session.h:
12798 media: enable per factory authorisations
12799 Allow for adding a GstRTSPAuth on the factory and media level and check
12800 permissions when accessing the factory.
12801 Add hints to the auth methods for future more fine grained authorisation.
12802 Add example application for per factory authentication.
12804 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12806 * gst/rtsp-server/rtsp-auth.c:
12807 * gst/rtsp-server/rtsp-auth.h:
12808 * gst/rtsp-server/rtsp-client.c:
12809 * gst/rtsp-server/rtsp-client.h:
12810 * gst/rtsp-server/rtsp-params.c:
12811 * gst/rtsp-server/rtsp-params.h:
12812 rtsp-server: Pass ClientState structure arround
12813 Pass the collected information for the ongoing request in a GstRTSPClientState
12814 structure that we can then pass around to simplify the method arguments. This
12815 will also be handy when we implement logging functionality.
12817 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12819 * gst/rtsp-server/rtsp-media-factory.c:
12820 * gst/rtsp-server/rtsp-media-factory.h:
12821 media-factory: add methods to configure authorisation
12823 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12825 * gst/rtsp-server/rtsp-client.c:
12826 client: unref auth in finalize
12828 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12830 * gst/rtsp-server/rtsp-server.c:
12831 server: unref auth in finalize
12833 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12835 * docs/libs/gst-rtsp-server-docs.sgml:
12836 * docs/libs/gst-rtsp-server-sections.txt:
12837 * docs/libs/gst-rtsp-server.types:
12838 docs: add more docs
12840 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12842 * gst/rtsp-server/rtsp-server.c:
12843 * gst/rtsp-server/rtsp-server.h:
12844 server: separate create and accept
12845 Create separate create and accept methods so that subclasses can create custom
12847 Configure the server in the client object and prepare for keeping track of
12850 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12852 * gst/rtsp-server/rtsp-client.c:
12853 * gst/rtsp-server/rtsp-client.h:
12854 client: add support for setting the server.
12855 Add support for keeping a ref to the server that started this client
12858 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12860 * gst/rtsp-server/rtsp-auth.c:
12861 auth: fix memleak and add some docs
12862 Fix a memleak of the basic auth token.
12863 Add docs for the helper function
12865 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12867 * gst/rtsp-server/rtsp-auth.c:
12868 * gst/rtsp-server/rtsp-auth.h:
12869 * gst/rtsp-server/rtsp-client.c:
12870 client: delegate setup of auth to the manager
12871 Delegate the configuration of the authentication tokens to the manager object
12874 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12876 * examples/test-video.c:
12877 * gst/rtsp-server/Makefile.am:
12878 * gst/rtsp-server/rtsp-auth.c:
12879 * gst/rtsp-server/rtsp-auth.h:
12880 * gst/rtsp-server/rtsp-client.c:
12881 * gst/rtsp-server/rtsp-client.h:
12882 * gst/rtsp-server/rtsp-server.c:
12883 * gst/rtsp-server/rtsp-server.h:
12884 auth: add authentication object
12885 Add an object that can check the authorization of requests.
12886 Implement basic authentication.
12887 Add example authentication to test-video
12889 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12891 * gst/rtsp-server/rtsp-server.c:
12892 * gst/rtsp-server/rtsp-server.h:
12893 server: move includes back
12894 the includes are needed for sockaddr_in.
12896 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12898 * gst/rtsp-server/rtsp-client.c:
12899 * gst/rtsp-server/rtsp-client.h:
12900 * gst/rtsp-server/rtsp-server.c:
12901 * gst/rtsp-server/rtsp-server.h:
12902 rtsp: move network includes where they are needed
12904 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12906 * gst/rtsp-server/rtsp-media.h:
12907 rtsp-media.h: Minor corrections in comments.
12910 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12913 Automatic update of common submodule
12914 From e572c87 to f94d739
12916 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12920 * docs/libs/.gitignore:
12921 * examples/.gitignore:
12922 * gst/rtsp-server/.gitignore:
12925 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12927 * docs/libs/Makefile.am:
12928 docs: We don't build ps/pdf for API reference docs
12930 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12933 Automatic update of common submodule
12934 From ccbaa85 to e572c87
12936 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12939 Automatic update of common submodule
12940 From 46445ad to ccbaa85
12942 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12944 * gst/rtsp-server/Makefile.am:
12945 * gst/rtsp-server/rtsp-funnel.c:
12946 * gst/rtsp-server/rtsp-funnel.h:
12947 * gst/rtsp-server/rtsp-media.c:
12948 funnel: rename fsfunnel to rtspfunnel
12949 Rename the funnel to avoid conflicts with the farsight one.
12951 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12953 * gst/rtsp-server/Makefile.am:
12954 * gst/rtsp-server/fs-funnel.c:
12955 * gst/rtsp-server/fs-funnel.h:
12956 * gst/rtsp-server/rtsp-media.c:
12957 rtsp-media: add and use fsfunnel
12958 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12959 select-all property that we need.
12961 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12963 * gst/rtsp-server/Makefile.am:
12964 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12965 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12966 for the g-ir-compiler, rather than just assuming the env var has
12969 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12976 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12978 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12981 * gst/rtsp-server/Makefile.am:
12982 gobject-introspection: fix g-i build for uninstalled setup
12983 Requires gst-plugins-base git (> 0.10.31.2).
12985 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12987 * examples/test-uri.c:
12988 examples: add some more options and comments
12990 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12992 * gst/rtsp-server/rtsp-media-factory-uri.c:
12993 factory-uri: use right property type
12995 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12997 * gst/rtsp-server/rtsp-media-factory-uri.c:
12998 factory-uri: attempt to configure buffer-lists
12999 Attempt to configure buffer lists in the payloader for improved performance.
13001 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13003 * gst/rtsp-server/rtsp-media.c:
13004 media: attempt to configure bigger UDP buffers
13005 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
13006 send buffers with high bitrate streams.
13008 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
13010 * gst/rtsp-server/rtsp-client.c:
13011 client: use the socket length from getsockname
13012 Use the length returned by getsockname to perform the getnameinfo call because
13013 the size can depend on the socket type and platform.
13016 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13018 * docs/libs/gst-rtsp-server-docs.sgml:
13019 * docs/libs/gst-rtsp-server-sections.txt:
13020 docs: add uri factory to the docs
13022 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13024 * gst/rtsp-server/rtsp-client.c:
13025 * gst/rtsp-server/rtsp-media.h:
13028 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13030 * gst/rtsp-server/rtsp-client.c:
13031 * gst/rtsp-server/rtsp-media.c:
13032 * gst/rtsp-server/rtsp-media.h:
13033 * gst/rtsp-server/rtsp-session.c:
13034 * gst/rtsp-server/rtsp-session.h:
13035 rtsp-server: add support for buffer lists
13036 Add support for sending bufferlists received from appsink.
13039 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13041 * gst/rtsp-server/rtsp-client.c:
13042 * gst/rtsp-server/rtsp-media.c:
13043 * gst/rtsp-server/rtsp-media.h:
13044 * gst/rtsp-server/rtsp-sdp.c:
13045 media: make method to retrieve the play range
13046 Make a method to retrieve the playback range so that we can conditionally create
13047 a different range for the SDP and the PLAY requests.
13049 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13051 * gst/rtsp-server/rtsp-media.c:
13052 * gst/rtsp-server/rtsp-media.h:
13053 media: add signal to notify of state changes
13055 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13057 * gst/rtsp-server/rtsp-client.h:
13058 client: cleanup headers
13060 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13062 * gst/rtsp-server/rtsp-client.c:
13065 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13067 * gst/rtsp-server/rtsp-media-factory-uri.c:
13068 * gst/rtsp-server/rtsp-media-factory-uri.h:
13069 factory-uri: add support for gstpay
13070 Add an option to prefer gstpay over decoder + raw payloader.
13072 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13074 * gst/rtsp-server/rtsp-media-factory-uri.c:
13075 * gst/rtsp-server/rtsp-media-factory-uri.h:
13076 factory-uri: rework the autoplugger.
13077 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
13080 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13082 * gst/rtsp-server/rtsp-media-factory-uri.c:
13083 factory-uri: use better factory filter
13084 Make better payloader filter based on autoplug rank and RTP use case.
13086 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13089 Automatic update of common submodule
13090 From 169462a to 46445ad
13092 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13094 * gst/rtsp-server/rtsp-server.c:
13095 server: set SO_REUSEADDR before bind
13096 Set the SO_REUSEADDR _before_ bind() to make it actually work.
13098 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13100 * gst/rtsp-server/rtsp-media.c:
13101 * gst/rtsp-server/rtsp-media.h:
13102 media: emit prepared signal when prepared
13103 Make a 'prepared' signal and emit it when we successfully prepared the element.
13104 This signal can be used to configure the media object after it has been prepared
13107 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
13110 Automatic update of common submodule
13111 From 011bcc8 to 169462a
13113 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
13115 python an optional dependency
13116 * configure.ac: Move up valgrind and g-i checks. Make the python
13117 dependency optional, as it was before.
13119 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13121 Merge branch 'master' into 0.11
13126 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13128 * gst/rtsp-server/rtsp-media.c:
13129 media: update range when active clients changed
13130 When we changed the number of active clients, update the current range
13131 information because we want the second client connecting to a shared resource
13132 continue from where the stream currently.
13134 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13136 * gst/rtsp-server/rtsp-media-factory-uri.c:
13137 * gst/rtsp-server/rtsp-media-factory-uri.h:
13138 factory-uri: add colorspace and fix pt
13139 Rework the way we pass data to the autoplugger.
13140 When we have raw caps, plug a converter element to make pluggin to raw
13141 payloaders more successful.
13142 Make sure all dynamically plugged payloaders have a unique payload types.
13144 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13146 * examples/Makefile.am:
13147 * examples/test-uri.c:
13148 example: add example of the uri factory
13150 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13152 * gst/rtsp-server/Makefile.am:
13153 * gst/rtsp-server/rtsp-media-factory-uri.c:
13154 * gst/rtsp-server/rtsp-media-factory-uri.h:
13155 * gst/rtsp-server/rtsp-server.h:
13156 factory-uri: add a factory to stream any URI
13157 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
13160 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13162 * gst/rtsp-server/rtsp-media.c:
13163 * gst/rtsp-server/rtsp-media.h:
13164 media: ignore spurious ASYNC_DONE messages
13165 When we are dynamically adding pads, the addition of the udpsrc elements will
13166 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
13167 the real ASYNC_DONE when everything is prerolled.
13169 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13171 * gst/rtsp-server/rtsp-media-factory.c:
13172 * gst/rtsp-server/rtsp-media-factory.h:
13173 media-factory: make lock macro
13175 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
13177 * gst/rtsp-server/rtsp-client.c:
13178 rtsp-server: Remove unused variable and dead assignment
13180 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
13182 * examples/test-launch.c:
13183 * examples/test-mp4.c:
13184 * examples/test-ogg.c:
13185 * examples/test-readme.c:
13186 * examples/test-sdp.c:
13187 * examples/test-video.c:
13188 examples: Run gst-indent
13190 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
13192 * gst/rtsp-server/rtsp-client.c:
13193 * gst/rtsp-server/rtsp-media-factory.c:
13194 * gst/rtsp-server/rtsp-media-mapping.c:
13195 * gst/rtsp-server/rtsp-media.c:
13196 * gst/rtsp-server/rtsp-params.c:
13197 * gst/rtsp-server/rtsp-sdp.c:
13198 * gst/rtsp-server/rtsp-server.c:
13199 * gst/rtsp-server/rtsp-session-pool.c:
13200 * gst/rtsp-server/rtsp-session.c:
13201 rtsp-server: Run gst-indent
13202 Since it wasn't using the upstream common previously, there was no
13203 indentation check before commiting.
13205 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
13207 * gst/rtsp-server/rtsp-media-mapping.h:
13208 * gst/rtsp-server/rtsp-media.c:
13209 * gst/rtsp-server/rtsp-media.h:
13210 * gst/rtsp-server/rtsp-sdp.c:
13211 * gst/rtsp-server/rtsp-session-pool.h:
13212 * gst/rtsp-server/rtsp-session.c:
13213 * gst/rtsp-server/rtsp-session.h:
13214 rtsp-server: Some more doc fixups
13216 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13219 Makefile: Add cruft-cleaning support
13221 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13225 * docs/Makefile.am:
13226 * docs/libs/Makefile.am:
13227 * docs/libs/gst-rtsp-server-docs.sgml:
13228 * docs/libs/gst-rtsp-server-sections.txt:
13229 * docs/libs/gst-rtsp-server.types:
13230 * docs/version.entities.in:
13231 docs: Add gtk-doc build system
13233 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13235 * gst/rtsp-server/Makefile.am:
13236 Makefile.am: Use standard GIR make behaviour
13238 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13242 autogen/configure: Bring more in sync to standard gst module behaviour
13244 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13246 * gst/rtsp-server/rtsp-media.c:
13247 media: warn and fail when gstrtpbin is not found
13249 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13252 configure: open 0.11 branch
13254 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
13258 Add common submodule
13260 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
13262 * common/ChangeLog:
13263 * common/Makefile.am:
13264 * common/c-to-xml.py:
13265 * common/check.mak:
13266 * common/coverage/coverage-report-entry.pl:
13267 * common/coverage/coverage-report.pl:
13268 * common/coverage/coverage-report.xsl:
13269 * common/coverage/lcov.mak:
13270 * common/gettext.patch:
13271 * common/glib-gen.mak:
13272 * common/gst-autogen.sh:
13273 * common/gst-xmlinspect.py:
13275 * common/gstdoc-scangobj:
13276 * common/gtk-doc-plugins.mak:
13277 * common/gtk-doc.mak:
13278 * common/m4/.gitignore:
13279 * common/m4/Makefile.am:
13280 * common/m4/README:
13281 * common/m4/as-ac-expand.m4:
13282 * common/m4/as-auto-alt.m4:
13283 * common/m4/as-compiler-flag.m4:
13284 * common/m4/as-compiler.m4:
13285 * common/m4/as-docbook.m4:
13286 * common/m4/as-libtool-tags.m4:
13287 * common/m4/as-libtool.m4:
13288 * common/m4/as-python.m4:
13289 * common/m4/as-scrub-include.m4:
13290 * common/m4/as-version.m4:
13291 * common/m4/ax_create_stdint_h.m4:
13292 * common/m4/check.m4:
13293 * common/m4/glib-gettext.m4:
13294 * common/m4/gst-arch.m4:
13295 * common/m4/gst-args.m4:
13296 * common/m4/gst-check.m4:
13297 * common/m4/gst-debuginfo.m4:
13298 * common/m4/gst-default.m4:
13299 * common/m4/gst-doc.m4:
13300 * common/m4/gst-error.m4:
13301 * common/m4/gst-feature.m4:
13302 * common/m4/gst-function.m4:
13303 * common/m4/gst-gettext.m4:
13304 * common/m4/gst-glib2.m4:
13305 * common/m4/gst-libxml2.m4:
13306 * common/m4/gst-plugindir.m4:
13307 * common/m4/gst-valgrind.m4:
13308 * common/m4/gtk-doc.m4:
13309 * common/m4/introspection.m4:
13310 * common/m4/pkg.m4:
13311 * common/mangle-tmpl.py:
13312 * common/plugins.xsl:
13314 * common/release.mak:
13315 * common/scangobj-merge.py:
13316 * common/upload.mak:
13317 common: Remove static version
13319 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
13321 * common/m4/introspection.m4:
13322 Update introspection.m4 to match usage
13324 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13328 Remove old stuff from the README
13330 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13333 back to development
13335 === release 0.10.7 ===
13337 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13342 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13344 * examples/test-ogg.c:
13345 test-ogg: remove parsers
13346 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
13347 buffers with timestamps. Using the parsers also seems to break things.
13349 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13351 * bindings/vala/gst-rtsp-server-0.10.vapi:
13352 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13353 Updated Vala bindings
13355 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13357 * common/m4/introspection.m4:
13359 * gst/rtsp-server/Makefile.am:
13360 Added initial gobject-introspection support
13362 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13364 * gst/rtsp-server/rtsp-media-factory.c:
13365 media-factory: don't use host for shared hash key
13366 When we generate the key to share made between connections, don't include the
13367 host used to connect so that we can share media even if between clients that
13368 connected with localhost and ones with the ip address.
13370 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13372 * bindings/vala/Makefile.am:
13373 build: fix distcheck
13375 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13377 * bindings/vala/gst-rtsp-server-0.10.vapi:
13378 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13379 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13380 Update Vala bindings
13382 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13384 * bindings/vala/Makefile.am:
13386 Fix configure checks and installation location for Vala bindings
13389 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13392 back to development
13394 === release 0.10.6 ===
13396 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13399 configure: release 0.10.6
13401 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13403 * gst/rtsp-server/rtsp-media.c:
13404 media: help the compiler a little
13406 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13408 * gst/rtsp-server/rtsp-media.c:
13409 * gst/rtsp-server/rtsp-media.h:
13410 * gst/rtsp-server/rtsp-session.c:
13411 media: cleanup media transport before freeing
13412 Cleanup the media transport data before freeing. In particular, remove the qdata
13413 from the rtpsource object.
13415 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13417 * gst/rtsp-server/rtsp-media-factory.c:
13418 * gst/rtsp-server/rtsp-media-factory.h:
13419 * gst/rtsp-server/rtsp-media.c:
13420 * gst/rtsp-server/rtsp-media.h:
13421 media-factory: add eos-shutdown property
13422 Add an eos-shutdown property that will send an EOS to the pipeline before
13423 shutting it down. This allows for nice cleanup in case of a muxer.
13426 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13428 * gst/rtsp-server/rtsp-media.c:
13429 * gst/rtsp-server/rtsp-media.h:
13430 media: use multiudpsink send-duplicates when we can
13431 If we have a new enough multiudpsink with the send-duplicates property, use this
13432 instead of doing our own filtering. Our custom filtering code should eventually
13433 be removed when we can depend on a released -good.
13435 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13437 * gst/rtsp-server/rtsp-media.c:
13438 media: don't leak destinations
13439 Refactor and cleanup the destinations array when the stream is destroyed.
13441 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13443 * gst/rtsp-server/rtsp-media.c:
13444 * gst/rtsp-server/rtsp-media.h:
13445 media: don't add udp addresses multiple times
13446 Keep track of the udp addresses we added to udpsink and never add the same udp
13447 destination twice. This avoids duplicate packets when using multicast.
13449 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13451 * gst/rtsp-server/rtsp-server.c:
13452 server: disable use of SO_LINGER
13453 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13454 server close()s the connection.
13456 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13458 * gst/rtsp-server/rtsp-server.c:
13459 server: use 5 second linger period in SO_LINGER
13460 Wait 5 seconds before clearing the send buffers and reseting the connection with
13461 the client when we do a close. This should be enough time to get the message to
13465 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13467 * gst/rtsp-server/rtsp-server.c:
13468 server: use SO_LINGER
13469 SO_LINGER on the socket will make sure that any pending data on the socket is
13470 flushed ASAP and that the socket connection is reset. This makes sure that the
13471 socket can be reused immediately.
13474 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13477 README: add blurb about shared media factories
13479 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13481 * gst/rtsp-server/rtsp-media.c:
13482 Add stdlib.h for atoi()
13484 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13486 * bindings/python/Makefile.am:
13487 * bindings/vala/Makefile.am:
13488 build: distcheck fixes
13489 Fix 'make distcheck', somewhat (it still fails because it tries to
13490 install files into /usr/share/vala/vapi/ irrespective of the
13491 configured prefix).
13493 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13496 configure: bump core/base requirements to released version
13497 Makes things less confusing for people.
13499 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13502 configure: fail if GStreamer core/base requirements are not met
13504 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13506 * gst/rtsp-server/rtsp-client.c:
13507 client: improve client cleanups
13508 Make sure the session does not timeout when using TCP. We need to do this
13509 because quicktime player does not send RTCP for some reason in tunneled
13511 Refactor some cleanup code.
13514 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13516 * gst/rtsp-server/rtsp-session.c:
13517 * gst/rtsp-server/rtsp-session.h:
13518 session: add support for prevent session timeouts
13519 Add an atomix counter to prevent session timeouts when we are, for example,
13520 streaming over TCP.
13522 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13524 * gst/rtsp-server/rtsp-client.c:
13525 client: fix unlink on session timeouts
13526 When our session times out, make sure we unlink all streams in this
13528 Remove the tunnelid when closing the connection.
13530 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13532 * gst/rtsp-server/rtsp-session.c:
13533 session: small cleanups
13535 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13537 * gst/rtsp-server/rtsp-client.c:
13538 client: handle lost_tunnel callbacks
13539 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13540 hashtable so that we can reuse it for when the client reopens the POST
13542 Close the connection after a TEARDOWN.
13543 Make sure or watchid is cleared when the watch is removed.
13546 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13548 * gst/rtsp-server/rtsp-client.c:
13549 * gst/rtsp-server/rtsp-media.c:
13550 * gst/rtsp-server/rtsp-sdp.c:
13551 rtsp-server: add more support for multicast
13553 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13556 * gst/rtsp-server/rtsp-media.c:
13557 * gst/rtsp-server/rtsp-media.h:
13558 media: allow configuration of allowed lower transport
13560 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13562 * gst/rtsp-server/rtsp-client.h:
13563 * gst/rtsp-server/rtsp-media.c:
13564 * gst/rtsp-server/rtsp-media.h:
13565 * gst/rtsp-server/rtsp-sdp.c:
13566 * gst/rtsp-server/rtsp-sdp.h:
13567 * gst/rtsp-server/rtsp-server.c:
13568 rtsp: keep track of server ip and ipv6
13569 Keep track of how the client connected to the server and setup the udp ports
13570 with the same protocol.
13571 Copy the server ip address in the SDP so that clients can send RTCP back to
13574 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13576 * gst/rtsp-server/rtsp-session.c:
13579 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13581 * gst/rtsp-server/rtsp-client.c:
13582 client: use right size for malloc
13584 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13586 * gst/rtsp-server/rtsp-server.c:
13587 server: comment ipv6 server listening address
13589 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13591 * gst/rtsp-server/rtsp-media.c:
13592 media: allow for ipv6 sockets
13594 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13596 * gst/rtsp-server/rtsp-server.c:
13597 * gst/rtsp-server/rtsp-server.h:
13598 server: rework server part
13599 Allow setting a bind address, make sure we can deal with ipv6.
13600 Remove the port property and change with the service property.
13602 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13604 * gst/rtsp-server/rtsp-media.h:
13605 media: update comments a little
13607 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13609 * gst/rtsp-server/rtsp-client.c:
13610 client: make content-base better
13611 Use the URI formatting functions to make a content-base. Also make sure that
13612 there is a trailing / at the end.
13614 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13616 * gst/rtsp-server/rtsp-client.c:
13617 client: guard against invalid paths
13619 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13621 * examples/test-video.c:
13622 test: catch server bind errors
13624 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13626 * gst/rtsp-server/rtsp-media.c:
13627 rtspmedia: emit "unprepared" if _prepare fails.
13628 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13629 media object is removed from its factory's cache.
13631 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13633 * gst/rtsp-server/rtsp-media.c:
13634 media: collect media position when seek completes
13636 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13638 * gst/rtsp-server/rtsp-client.c:
13639 client: call unlink_streams in client finalize
13642 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13644 * gst/rtsp-server/rtsp-media.c:
13645 media: limit the time to wait to something huge
13646 Avoid waiting forever but limit the timeout to 20 seconds.
13648 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13650 * gst/rtsp-server/rtsp-sdp.c:
13651 sdp: reindent and check for prepared status
13653 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13655 * gst/rtsp-server/rtsp-media.c:
13656 * gst/rtsp-server/rtsp-media.h:
13657 * gst/rtsp-server/rtsp-session.c:
13658 media: avoid doing _get_state() for state changes
13659 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13660 until the media is prerolled or in error. This avoids doing a blocking call of
13661 gst_element_get_state() that can cause lockups when there is an error.
13664 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13666 * gst/rtsp-server/rtsp-media.c:
13669 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13671 * gst/rtsp-server/rtsp-media-factory.c:
13672 media-factory: better error handling
13673 Improve the error handling a bit.
13675 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13677 * gst/rtsp-server/rtsp-client.c:
13678 client: rework transport parsing
13679 Rework the transport parsing code so that we can ignore transports we don't
13680 support instead of just picking the first one we can parse.
13681 Configure a (for now hardcoded) destination for multicast transports.
13683 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13685 * gst/rtsp-server/rtsp-media.c:
13686 media: set multicast sink parameters
13687 Disable loop and automatic multicast join on the udpsink elements.
13688 Add some more debug info.
13689 Reset some state variables in the right place.
13690 Use the right port numbers for multicast.
13692 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13694 * gst/rtsp-server/rtsp-session.c:
13695 session: handle transport setup correctly
13696 Handle UDP, MCAST and TCP transport negotiation more correctly.
13697 Store the server session SSRC in the transport.
13699 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13701 * gst/rtsp-server/rtsp-client.c:
13702 rtsp-client: implement error_full
13703 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13706 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13709 * gst/rtsp-server/rtsp-client.c:
13710 * gst/rtsp-server/rtsp-server.c:
13711 docs: update docs and comments
13713 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13715 * gst/rtsp-server/rtsp-sdp.c:
13716 sdp: make server work better when behind a proxy
13718 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13720 * gst/rtsp-server/rtsp-client.c:
13721 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13723 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13725 * gst/rtsp-server/rtsp-client.c:
13726 * gst/rtsp-server/rtsp-media-factory.c:
13727 * gst/rtsp-server/rtsp-media-mapping.c:
13728 * gst/rtsp-server/rtsp-media.c:
13729 * gst/rtsp-server/rtsp-server.c:
13730 * gst/rtsp-server/rtsp-session-pool.c:
13731 * gst/rtsp-server/rtsp-session.c:
13732 Use GStreamer's debugging subsystem
13734 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13736 * gst/rtsp-server/rtsp-media-factory.c:
13737 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13739 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13742 back to development
13744 === release 0.10.5 ===
13746 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13751 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13754 configure: bump required versions
13756 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13758 * gst/rtsp-server/rtsp-client.c:
13759 client: call weak-unref on client->sessions from finalize
13762 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13764 * gst/rtsp-server/rtsp-media.c:
13765 media: Fixed crasher where caps got unref'ed too often
13767 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13770 * pkgconfig/.gitignore:
13771 * pkgconfig/Makefile.am:
13772 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13773 Added pkg-config file to use gst-rtsp-server uninstalled
13775 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13777 * gst/rtsp-server/rtsp-media.c:
13778 media: add some docs
13780 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13782 * gst/rtsp-server/rtsp-client.c:
13783 rtsp: Use gst_rtsp_watch_send_message().
13784 Use gst_rtsp_watch_send_message() since the old API which used
13785 gst_rtsp_watch_queue_message() has been deprecated.
13787 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13790 back to development
13792 === release 0.10.4 ===
13794 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13799 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13801 * gst/rtsp-server/rtsp-client.c:
13802 * gst/rtsp-server/rtsp-session.c:
13803 * gst/rtsp-server/rtsp-session.h:
13804 rtsp: allocate channels in TCP mode
13805 When the client does not provide us with channels in TCP mode, allocate channels
13808 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13810 * gst/rtsp-server/rtsp-client.c:
13811 client: don't crash when tunnelid is missing
13812 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13813 don't crash but return an error response to the client.
13816 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13818 * bindings/vala/gst-rtsp-server-0.10.vapi:
13819 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13820 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13821 bindings: update vala bindings with new method
13823 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13825 * gst/rtsp-server/rtsp-session-pool.c:
13826 * gst/rtsp-server/rtsp-session-pool.h:
13827 sessionpool: add function to filter sessions
13828 Add generic function to retrieve/remove sessions.
13830 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13833 configure: bump core/base requirements to release
13835 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13837 * gst/rtsp-server/rtsp-media.c:
13838 media: fix indentation
13840 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13842 * gst/rtsp-server/rtsp-media.c:
13843 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13845 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13847 * gst/rtsp-server/rtsp-media.c:
13848 set state and remove elements of media in for loop
13850 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13852 * bindings/vala/gst-rtsp-server-0.10.vapi:
13853 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13854 Added gst_rtsp_media_remove_elements function to Vala bindings
13856 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13858 * gst/rtsp-server/rtsp-media.c:
13859 * gst/rtsp-server/rtsp-media.h:
13860 Added gst_rtsp_media_remove_elements function
13862 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13864 * gst/rtsp-server/rtsp-media.c:
13865 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13867 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13869 * bindings/vala/gst-rtsp-server-0.10.vapi:
13870 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13871 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13872 Updated Vala bindings
13874 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13876 * gst/rtsp-server/rtsp-media.c:
13877 * gst/rtsp-server/rtsp-media.h:
13878 Added vmethod unprepare to GstRTSPMedia
13879 The default implementation sets the state of the pipeline to GST_STATE_NULL
13881 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13883 * gst/rtsp-server/rtsp-media-factory.c:
13884 * gst/rtsp-server/rtsp-media-factory.h:
13885 Made collect_streams function public
13887 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13889 * gst/rtsp-server/rtsp-media-factory.c:
13890 * gst/rtsp-server/rtsp-media-factory.h:
13891 * gst/rtsp-server/rtsp-media.c:
13892 Added vmethod create_pipeline to GstRTSPMediaFactory
13893 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13895 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13897 * gst/rtsp-server/rtsp-client.c:
13898 client: use g_source_destroy()
13899 We need to use g_source_destroy() because we might have added the source to a
13900 different main context than the default one.
13902 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13904 * gst/rtsp-server/Makefile.am:
13905 * gst/rtsp-server/rtsp-client.c:
13906 * gst/rtsp-server/rtsp-params.c:
13907 * gst/rtsp-server/rtsp-params.h:
13908 rtsp: prepare for handling GET/SET_PARAMETER
13909 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13911 Fix return codes of handlers.
13913 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13915 * gst/rtsp-server/rtsp-media.c:
13916 media: don't leak session pads
13918 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13920 * gst/rtsp-server/rtsp-media.c:
13921 media: clean up the messages a bit
13923 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13925 * gst/rtsp-server/rtsp-sdp.c:
13926 sdp: warn and skip streams without media
13928 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13930 * bindings/vala/gst-rtsp-server-0.10.vapi:
13931 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13932 vala: Fixed typo in header file of RTSPMediaStream
13934 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13936 * gst/rtsp-server/rtsp-media.c:
13938 Fix a debug message
13939 Make dumping RTCP stats configurable
13941 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13943 * gst/rtsp-server/rtsp-media.c:
13944 media: be less verbose and leak less
13946 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13948 * gst/rtsp-server/rtsp-media.c:
13949 media: don't leak the destination address
13951 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13953 * gst/rtsp-server/rtsp-client.c:
13954 * gst/rtsp-server/rtsp-media.c:
13955 * gst/rtsp-server/rtsp-media.h:
13956 * gst/rtsp-server/rtsp-session.c:
13957 * gst/rtsp-server/rtsp-session.h:
13958 rtsp: use RTCP to keep the session alive
13959 Use the RTCP rtcp-from stats field to find the associated session and use this
13960 to keep the session alive.
13962 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13964 * gst/rtsp-server/rtsp-session.c:
13965 session: add 5sec to the real session timeout
13966 Allow the session to live 5sec longer before really timing out. This should give
13967 clients some extra time to keep the session active.
13969 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13971 * gst/rtsp-server/rtsp-client.c:
13972 client: replay OK to GET/SET_PARAMETER
13973 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13974 so that we return OK for those requests.
13976 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13978 * gst/rtsp-server/rtsp-media.c:
13979 * gst/rtsp-server/rtsp-media.h:
13980 media: keep track of active transports
13981 Keep track of which transport is active to avoid closing the connection too
13983 Remove the destination transport also when going to NULL.
13984 Print some stats about the SDES and other RTCP messages we receive from the
13987 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13989 * examples/.gitignore:
13990 * examples/Makefile.am:
13991 * examples/test-sdp.c:
13992 example: add SDP relay example
13994 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13996 * gst/rtsp-server/rtsp-media.c:
13997 media: also count active TCP connections
13999 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14001 * gst/rtsp-server/rtsp-media-factory.c:
14002 * gst/rtsp-server/rtsp-media.c:
14003 * gst/rtsp-server/rtsp-media.h:
14004 rtsp: add support for dynamic elements
14005 Add support for dynamic elements.
14006 Don't set live pipelines back to paused.
14008 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14010 * gst/rtsp-server/rtsp-sdp.c:
14011 sdp: don't add encoding name when absent in caps
14013 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14015 * gst/rtsp-server/rtsp-client.c:
14016 client: warn when we can't do RTP-Info
14018 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14020 * gst/rtsp-server/rtsp-media-factory.c:
14021 factory: factor out the stream construction
14023 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14025 * gst/rtsp-server/rtsp-client.c:
14026 client: only add RTP-Info when we have the info
14027 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
14030 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14033 back to development
14035 === release 0.10.3 ===
14037 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14041 - Fixes a bug where it put the wrong verion in pkgconfig
14042 - Link RTP and RTCP sources
14044 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14046 * gst/rtsp-server/rtsp-media.c:
14047 * gst/rtsp-server/rtsp-media.h:
14048 media: link the RTP udpsrc to the session manager
14049 Link the RTP udpsrc and the appsrc to the session manager so that they don't
14050 shut down when the client sends a packet to open firewalls.
14052 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14054 * pkgconfig/gst-rtsp-server.pc.in:
14055 Don't use hard-coded version number in pkg-config file
14057 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14060 back to development
14062 === release 0.10.2 ===
14064 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14069 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14072 * common/m4/.gitignore:
14073 * examples/.gitignore:
14074 * pkgconfig/.gitignore:
14075 add some .gitignore files
14077 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14079 * gst/rtsp-server/rtsp-media.c:
14080 media: seek to key frames
14082 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14084 * gst/rtsp-server/rtsp-media.c:
14085 media: emit the unprepared signal by id
14086 Emit the unprepared signal by id instead of name and set the media as
14089 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14091 * gst/rtsp-server/rtsp-media.c:
14092 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
14094 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14096 * gst/rtsp-server/rtsp-server.c:
14097 Added finalize function to GstRTPSPServer to unref session pool and media mapping
14099 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14101 * bindings/vala/gst-rtsp-server-0.10.vapi:
14102 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14103 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14104 Updated vala bindings
14106 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14108 * gst/rtsp-server/Makefile.am:
14109 * gst/rtsp-server/rtsp-client.c:
14110 * gst/rtsp-server/rtsp-media.c:
14111 server: use appsink and appsrc with the API
14112 Use the appsink/appsrc API instead of the signals for higher
14115 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14117 * examples/test-ogg.c:
14118 tests: set the payload type correctly
14120 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14122 * gst/rtsp-server/rtsp-media-factory.c:
14123 factory: connect to the unprepare signal
14124 Connect to the unprepare signal for non-reusable media so that we can remove
14125 them from the cache.
14127 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14129 * gst/rtsp-server/rtsp-media.c:
14130 * gst/rtsp-server/rtsp-media.h:
14131 media: add signal to notify of unprepare
14133 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14135 * gst/rtsp-server/rtsp-media.c:
14136 * gst/rtsp-server/rtsp-media.h:
14137 media: more work on making the media shared
14138 Add a reusable flag to medias, indicating that they can be reused after a state
14142 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14144 * examples/test-readme.c:
14145 examples: mark the example as shared for testing
14147 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14149 * gst/rtsp-server/rtsp-media.c:
14150 * gst/rtsp-server/rtsp-media.h:
14151 client: support shared media
14152 Always perform the state actions even if the target state of the pipeline is
14153 already correct, we still want to add/remove the transports when we are dealing
14155 Keep a counter of the number of active transports for a media so that we can use
14156 this to perform a state change when needed.
14157 Perform a state change of the pipeline only when the first transport was added
14158 or when there are no active transports.
14160 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14162 * gst/rtsp-server/rtsp-client.c:
14163 client: fix refcounting crasher
14164 Don't need to remove the weak refs in the finalize methods, they are already
14165 removed in the dispose.
14166 Don't register the callback with a DestroyNofity.
14168 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14170 * gst/rtsp-server/rtsp-client.c:
14171 Fix rtsp client refcount management in TCP mode.
14172 Don't unref a client ref we never had. Fixes an unref
14173 of an already-free client object after a client
14174 teardown request for me.
14176 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14178 * gst/rtsp-server/rtsp-session.c:
14179 docs: fix typo in API docs
14181 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14183 * gst/rtsp-server/rtsp-media.c:
14184 More seeking fixes.
14185 Keep the udp sources in playing even if we go to paused. unlock the sources when
14187 Add some more debug info.
14188 Only seek when we need to.
14189 Keep track of the position when we go to paused.
14191 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14193 * gst/rtsp-server/rtsp-client.c:
14194 * gst/rtsp-server/rtsp-media.c:
14195 * gst/rtsp-server/rtsp-media.h:
14196 Add beginnings of seeking.
14197 Parse the Range header and perform a seek on the pipeline for the requested
14198 position. It's disabled currently until I figure out what's going wrong.
14200 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14202 * gst/rtsp-server/rtsp-client.c:
14203 allow pause requests for now.
14206 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14208 * gst/rtsp-server/rtsp-client.c:
14209 Remove weak ref on the session in teardown
14210 We need to remove our weakref from the session when we do a teardown because
14211 else we close the TCP connection prematurely.
14213 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14215 * gst/rtsp-server/rtsp-client.c:
14216 * gst/rtsp-server/rtsp-client.h:
14217 * gst/rtsp-server/rtsp-session-pool.c:
14218 Do some more session cleanup
14219 Make session timeout kill the TCP connection that currently watches the
14221 Remove the client timeout property.
14223 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14225 * gst/rtsp-server/rtsp-client.c:
14226 * gst/rtsp-server/rtsp-client.h:
14227 * gst/rtsp-server/rtsp-media.c:
14228 * gst/rtsp-server/rtsp-media.h:
14229 * gst/rtsp-server/rtsp-server.c:
14230 * gst/rtsp-server/rtsp-session.c:
14231 * gst/rtsp-server/rtsp-session.h:
14233 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
14236 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14238 * examples/Makefile.am:
14239 * examples/test-launch.c:
14240 Add example server that takes launch lines
14241 Add an example server that streams any -launch line.
14243 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14245 * examples/test-readme.c:
14246 * gst/rtsp-server/rtsp-client.c:
14247 * gst/rtsp-server/rtsp-media.c:
14248 * gst/rtsp-server/rtsp-media.h:
14249 Add support for live streams
14250 Add support for live streams and ranges
14251 Start on handling TCP data transfer.
14253 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14255 * gst/rtsp-server/rtsp-media.c:
14256 Free the pipeline before other things
14259 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14261 * gst/rtsp-server/rtsp-client.c:
14262 Only free the pending tunnel if there is one
14265 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14267 * gst/rtsp-server/rtsp-client.c:
14268 * gst/rtsp-server/rtsp-client.h:
14269 * gst/rtsp-server/rtsp-media.c:
14270 rtsp-server: Add support for tunneling
14271 Add support for tunneling over HTTP.
14272 Use new connection methods to retrieve the url.
14273 Dispatch messages based on the message type instead of blindly
14274 assuming it's always a request.
14275 Keep track of the watch id so that we can remove it later.
14276 Set the media pipeline to NULL before unreffing the pipeline.
14278 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14280 * gst/rtsp-server/rtsp-client.c:
14281 * gst/rtsp-server/rtsp-client.h:
14282 Fix for channel -> watch rename in gstreamer
14283 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
14285 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14287 * gst/rtsp-server/rtsp-client.c:
14288 * gst/rtsp-server/rtsp-client.h:
14290 Use the async RTSP channels instead of spawning a new thread for each client.
14291 If a sessionid is specified in a request, fail if we don't have the session.
14293 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14295 * gst/rtsp-server/rtsp-media.c:
14296 Add better debug info
14297 Add some better debug info.
14299 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14301 * examples/test-video.c:
14303 Add support for session timeouts in the example.
14305 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14307 * gst/rtsp-server/rtsp-session-pool.c:
14308 * gst/rtsp-server/rtsp-session-pool.h:
14309 Pass GTimeVal around for performance reasons
14310 Get the current time only once and pass it around so that sessions don't have to
14311 get the current time anymore.
14312 Add experimental support for a GSource that dispatches when the session needs to
14315 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14317 * gst/rtsp-server/rtsp-session.c:
14318 * gst/rtsp-server/rtsp-session.h:
14319 Add better support for session timeouts
14320 Add a method to request the number of milliseconds when a session will timeout.
14322 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14324 * gst/rtsp-server/rtsp-media.c:
14325 * gst/rtsp-server/rtsp-media.h:
14326 Add suport for RTP manager monitoring
14327 Add the first stage in monitoring the rtp manager.
14328 Make sure we don't update the state to something we don't want.
14330 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14332 * gst/rtsp-server/rtsp-client.c:
14333 Add support for session keepalive
14334 Get and update the session timeout for all requests. get the session as early as
14337 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14339 * gst/rtsp-server/rtsp-media-factory.h:
14340 * gst/rtsp-server/rtsp-media.c:
14341 * gst/rtsp-server/rtsp-media.h:
14342 Handle media bus messages
14343 Handle media bus messages in a custom mainloop and dispatch them to the
14344 RTSPMedia objects. Let the default implementation handle some common messages.
14346 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14348 * gst/rtsp-server/rtsp-client.c:
14349 * gst/rtsp-server/rtsp-session-pool.c:
14350 * gst/rtsp-server/rtsp-session.c:
14351 Some more session timeout handling
14352 Move the session header setting code to a central place so that we always add
14353 the timeout parameter too.
14354 Handle timeouts by running the session cleanup code.
14355 Stop media before cleaning up.
14357 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14359 * gst/rtsp-server/rtsp-client.c:
14360 * gst/rtsp-server/rtsp-client.h:
14361 Add timeout property
14362 Add a timeout property ot the client and make the other properties into GObject
14365 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14367 * gst/rtsp-server/rtsp-session-pool.c:
14368 Use getters and setters in property code
14369 Use the getters and setters for the timeout property instead of locking
14372 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14374 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
14376 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14378 * gst/rtsp-server/rtsp-session-pool.c:
14379 * gst/rtsp-server/rtsp-session-pool.h:
14380 * gst/rtsp-server/rtsp-session.c:
14381 * gst/rtsp-server/rtsp-session.h:
14382 Add more timeout stuff
14383 Add method to check if a session is expired.
14384 Add method to perform cleanup on a session pool.
14386 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14388 * gst/rtsp-server/rtsp-client.c:
14389 * gst/rtsp-server/rtsp-session-pool.c:
14390 * gst/rtsp-server/rtsp-session-pool.h:
14391 * gst/rtsp-server/rtsp-session.c:
14392 * gst/rtsp-server/rtsp-session.h:
14393 Add beginnings of session timeouts and limits
14394 Add the timeout value to the Session header for unusual timeout values.
14395 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14396 limit on the amount of retry we do after a sessionid collision.
14397 Add properties to the sessionid and the timeout of a session. Keep track of
14398 creation time and last access time for sessions.
14400 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14402 * gst/rtsp-server/rtsp-client.c:
14403 * gst/rtsp-server/rtsp-media.c:
14404 * gst/rtsp-server/rtsp-media.h:
14405 * gst/rtsp-server/rtsp-sdp.c:
14406 * gst/rtsp-server/rtsp-session-pool.c:
14407 * gst/rtsp-server/rtsp-session.c:
14408 * gst/rtsp-server/rtsp-session.h:
14409 Cleanup of sessions and more
14410 Fix the refcounting of media and sessions in the client. Properly clean up the
14411 session data when the client performs a teardown.
14412 Add Server header to responses.
14413 Allow for multiple uri setups in one session.
14414 Add Range header to the PLAY response and add the range attribute to the SDP
14416 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14417 give the ownership of the sessionid to the session object.
14419 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14421 * gst/rtsp-server/rtsp-server.c:
14422 * gst/rtsp-server/rtsp-server.h:
14424 Rename the 'server_port' variable to simply 'port'.
14426 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14429 * gst/rtsp-server/rtsp-client.c:
14430 * gst/rtsp-server/rtsp-media.c:
14431 * gst/rtsp-server/rtsp-media.h:
14432 * gst/rtsp-server/rtsp-session.c:
14433 * gst/rtsp-server/rtsp-session.h:
14434 Rework the way we handle transports for streams
14435 Make the media accept an array of transports for the streams that we have
14436 configured for the play/pause requests.
14437 Implement server states for a client and its media.
14438 Require 0.10.22.1 (git HEAD) of gstreamer.
14440 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14442 * gst/rtsp-server/rtsp-client.c:
14443 * gst/rtsp-server/rtsp-media-factory.c:
14444 Drop const from functions dealing with urls
14445 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14446 have the right const in them.
14448 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14450 * gst/rtsp-server/rtsp-client.c:
14451 * gst/rtsp-server/rtsp-media.c:
14452 * gst/rtsp-server/rtsp-sdp.c:
14456 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14458 * gst/rtsp-server/rtsp-client.c:
14459 * gst/rtsp-server/rtsp-media-factory.c:
14460 * gst/rtsp-server/rtsp-media.c:
14461 * gst/rtsp-server/rtsp-media.h:
14463 Don't keep a reference to the GstRTSPMedia in the stream.
14464 Free more things when freeing the GstRTSPMedia.
14466 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14469 * gst/rtsp-server/rtsp-media-factory.c:
14470 * gst/rtsp-server/rtsp-media-factory.h:
14471 * gst/rtsp-server/rtsp-media.c:
14472 * gst/rtsp-server/rtsp-media.h:
14473 * gst/rtsp-server/rtsp-server.c:
14474 * gst/rtsp-server/rtsp-server.h:
14475 More docs and small cleanups
14476 Add some more docs and update the README
14477 Cleanup some method names.
14478 Remove an unneeded idx field in the GstRTSPMediaStream
14480 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14483 * examples/Makefile.am:
14484 * examples/test-readme.c:
14485 Add a README and more example code
14486 Add a README file that contains a small introduction on how to use the server
14487 along with the example code explained in the readme.
14489 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14491 * gst/rtsp-server/rtsp-media.c:
14492 * gst/rtsp-server/rtsp-server.c:
14493 Fix some leaks and change default port
14494 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14495 we finished the initial preroll. If we keep them locked, setting the pipeline to
14496 NULL will not stop and clean up the sources correctly.
14497 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14499 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14501 * gst/rtsp-server/rtsp-session.c:
14502 * gst/rtsp-server/rtsp-session.h:
14503 Cleanups to the session object
14504 Remove some unneeded variables in the session state of a stream such as the
14505 owner media and the server transport.
14506 Get the configuration of a media stream in a session based on the media_stream
14507 in the original object instead of our cached index.
14508 Free more data in the finalize method.
14510 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14512 * gst/rtsp-server/rtsp-client.c:
14513 * gst/rtsp-server/rtsp-client.h:
14514 Cleanups and reuse media from DESCRIBE
14515 Handle thread create errors.
14516 Rename some internal methods to better match what they actually do.
14517 Handle misconfiguration of session_pool and media_mapping gracefully.
14518 Cache the DESCRIBE media and uri in the client connection and reuse them when
14519 we receive a SETUP request in the same connection for the same uri.
14520 Cleanup the client connection object.
14522 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14524 * gst/rtsp-server/rtsp-media-factory.c:
14525 * gst/rtsp-server/rtsp-media-factory.h:
14526 * gst/rtsp-server/rtsp-media.c:
14527 * gst/rtsp-server/rtsp-media.h:
14528 Add shared properties to media and factory
14529 Add the shared property to media.
14530 Implement some simple caching in the factory depending on if the media is shared
14533 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14535 * gst/rtsp-server/rtsp-client.c:
14536 Add a little comment
14537 Add some comment about the content-base header.
14539 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14541 * examples/Makefile.am:
14542 * examples/test-mp4.c:
14543 * examples/test-ogg.c:
14544 * examples/test-video.c:
14545 * gst/rtsp-server/Makefile.am:
14546 * gst/rtsp-server/rtsp-client.c:
14547 * gst/rtsp-server/rtsp-client.h:
14548 * gst/rtsp-server/rtsp-media-factory.c:
14549 * gst/rtsp-server/rtsp-media-factory.h:
14550 * gst/rtsp-server/rtsp-media.c:
14551 * gst/rtsp-server/rtsp-media.h:
14552 * gst/rtsp-server/rtsp-sdp.c:
14553 * gst/rtsp-server/rtsp-sdp.h:
14554 * gst/rtsp-server/rtsp-server.c:
14555 * gst/rtsp-server/rtsp-server.h:
14556 * gst/rtsp-server/rtsp-session.c:
14557 * gst/rtsp-server/rtsp-session.h:
14558 Reorganize things, prepare for media sharing
14559 Added various other test server examples
14560 Move the SDP message generation to a separate helper.
14561 Refactor common code for finding the session.
14562 Add content-base for realplayer compatibility
14563 Clean up request uris before processing for better vlc compatibility.
14564 Move prerolling and pipeline construction to the RTSPMedia object.
14565 Use multiudpsink for future pipeline reuse.
14567 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14570 Back to development
14573 === release 0.10.1 ===
14575 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14578 Make 0.10.1 release
14581 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14583 * bindings/vala/Makefile.am:
14585 Add more directories and files to the dist.
14587 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14589 * bindings/python/Makefile.am:
14590 * bindings/python/rtspserver.override:
14591 Fixed compile error of python bindings
14593 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14595 * bindings/vala/gst-rtsp-server-0.10.vapi:
14596 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14597 Marked values as nullable accordingly
14599 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14601 * bindings/vala/gst-rtsp-server-0.10.vapi:
14602 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14603 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14604 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14605 Updated Vala bindings
14607 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14609 * gst/rtsp-server/rtsp-client.c:
14610 * gst/rtsp-server/rtsp-media-mapping.c:
14611 * gst/rtsp-server/rtsp-media-mapping.h:
14612 * gst/rtsp-server/rtsp-media.h:
14613 * gst/rtsp-server/rtsp-session-pool.h:
14614 Cleanups and doc updates
14615 Add some more documentation and do some minor cleanups here and there.
14617 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14619 * gst/rtsp-server/rtsp-client.c:
14620 * gst/rtsp-server/rtsp-media-factory.c:
14621 * gst/rtsp-server/rtsp-media-factory.h:
14622 * gst/rtsp-server/rtsp-media.c:
14623 * gst/rtsp-server/rtsp-media.h:
14624 * gst/rtsp-server/rtsp-session.c:
14625 * gst/rtsp-server/rtsp-session.h:
14627 Rename GstRTSPMediaBin to GstRTSPMedia
14628 Parse the request url into a GstRTSPUri object and pass this object to the
14629 various handlers and methods that require the uri.
14631 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14635 Add some more docs and remove some old code from the example.
14637 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14639 * gst/rtsp-server/rtsp-client.c:
14640 Handle state change failures better
14641 Handle state change failures better when changing the state of the pipeline to
14644 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14646 * gst/rtsp-server/rtsp-media-factory.c:
14647 * gst/rtsp-server/rtsp-media-factory.h:
14648 Make element creation more extendible
14649 Add get_element vmethod to the default MediaFactory so that subclasses can just
14650 override that method and still use the default logic for making a MediaBin from
14653 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14656 * gst/rtsp-server/Makefile.am:
14657 * gst/rtsp-server/rtsp-client.c:
14658 * gst/rtsp-server/rtsp-client.h:
14659 * gst/rtsp-server/rtsp-media-factory.c:
14660 * gst/rtsp-server/rtsp-media-factory.h:
14661 * gst/rtsp-server/rtsp-media-mapping.c:
14662 * gst/rtsp-server/rtsp-media-mapping.h:
14663 * gst/rtsp-server/rtsp-media.c:
14664 * gst/rtsp-server/rtsp-media.h:
14665 * gst/rtsp-server/rtsp-server.c:
14666 * gst/rtsp-server/rtsp-server.h:
14667 * gst/rtsp-server/rtsp-session.c:
14668 * gst/rtsp-server/rtsp-session.h:
14669 Make the server handle arbitrary pipelines
14670 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14671 The GstMediaBin object has a handle to a bin with elements and to a list of
14672 GstMediaStream objects that this bin produces.
14673 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14674 with methods to register and remove those mappings.
14675 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14676 used by the server instance.
14677 Modify the example application so that it shows how to create custom pipelines
14678 attached to a specific mount point.
14679 Various misc cleanps.
14681 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14683 * gst/rtsp-server/rtsp-server.c:
14684 * gst/rtsp-server/rtsp-server.h:
14685 Allow setting a custom media factory for a server
14687 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14689 * gst/rtsp-server/rtsp-client.c:
14690 * gst/rtsp-server/rtsp-client.h:
14691 Allow setting a custom media factory for a client.
14693 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14695 * gst/rtsp-server/Makefile.am:
14696 Add Makefile entry for the media factory
14698 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14700 * gst/rtsp-server/rtsp-media-factory.c:
14701 * gst/rtsp-server/rtsp-media-factory.h:
14702 Add media factory to map urls to media pipeline objects.
14704 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14706 * gst/rtsp-server/rtsp-media.c:
14707 * gst/rtsp-server/rtsp-media.h:
14708 Add comments. Remove unused field
14710 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14712 * gst/rtsp-server/rtsp-session-pool.c:
14713 * gst/rtsp-server/rtsp-session-pool.h:
14714 Allow custom session pools to override the session id allocation algorithms Add some comments.
14716 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14718 * gst/rtsp-server/rtsp-session.h:
14721 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14723 * gst/rtsp-server/rtsp-client.c:
14724 * gst/rtsp-server/rtsp-client.h:
14725 Move the connection code in one place Add some comments
14727 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14729 * gst/rtsp-server/rtsp-server.c:
14730 * gst/rtsp-server/rtsp-server.h:
14731 Make vmethod to create and accept new clients. Add some docs.
14733 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14735 * gst/rtsp-server/rtsp-server.c:
14736 * gst/rtsp-server/rtsp-server.h:
14737 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14739 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14741 * gst/rtsp-server/rtsp-client.c:
14742 * gst/rtsp-server/rtsp-client.h:
14743 Name the parameters more appropriately.
14745 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14747 * gst/rtsp-server/rtsp-session-pool.c:
14748 Do some more cleanup of the session pool.
14750 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14752 * gst/rtsp-server/Makefile.am:
14753 * gst/rtsp-server/rtsp-client.c:
14754 Check if return value of gst_rtsp_session_get_media is not NULL
14756 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14758 * gst/rtsp-server/Makefile.am:
14759 Install rtsp-session and rtsp-session-pool headers
14761 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14766 * bindings/python/Makefile.am:
14767 * bindings/python/arg-types.py:
14768 * bindings/python/codegen/Makefile.am:
14769 * bindings/python/codegen/__init__.py:
14770 * bindings/python/codegen/argtypes.py:
14771 * bindings/python/codegen/code-coverage.py:
14772 * bindings/python/codegen/codegen.py:
14773 * bindings/python/codegen/definitions.py:
14774 * bindings/python/codegen/defsparser.py:
14775 * bindings/python/codegen/docextract.py:
14776 * bindings/python/codegen/docgen.py:
14777 * bindings/python/codegen/fileprefix.override:
14778 * bindings/python/codegen/fileprefixmodule.c:
14779 * bindings/python/codegen/h2def.py:
14780 * bindings/python/codegen/mergedefs.py:
14781 * bindings/python/codegen/mkskel.py:
14782 * bindings/python/codegen/override.py:
14783 * bindings/python/codegen/reversewrapper.py:
14784 * bindings/python/codegen/scmexpr.py:
14785 * bindings/python/rtspserver-types.defs:
14786 * bindings/python/rtspserver.defs:
14787 * bindings/python/rtspserver.override:
14788 * bindings/python/rtspservermodule.c:
14790 Add python bindings.
14792 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14794 * bindings/Makefile.am:
14796 Don't go into python dir when requirements for python bindings are missing
14798 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14800 * bindings/Makefile.am:
14801 * bindings/vala/Makefile.am:
14803 Install Vala bindings if vala is available
14805 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14807 * bindings/vala/gst-rtsp-server-0.10.deps:
14808 * bindings/vala/gst-rtsp-server-0.10.vapi:
14809 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14810 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14811 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14812 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14813 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14814 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14815 Regenerated Vala bindings
14817 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14819 * bindings/vala/gst-rtsp-server.vapi:
14820 * bindings/vala/packages/gst-rtsp-server.metadata:
14821 Fixed typo in included headers for vala bindings
14823 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14827 * pkgconfig/Makefile.am:
14828 * pkgconfig/gst-rtsp-server.pc.in:
14829 Added pkgconfig file
14831 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14833 * bindings/vala/gst-rtsp-server.vapi:
14834 * bindings/vala/packages/gst-rtsp-server.excludes:
14835 * bindings/vala/packages/gst-rtsp-server.gi:
14836 * bindings/vala/packages/gst-rtsp-server.metadata:
14837 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14839 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14841 * bindings/vala/gst-rtsp-server.vapi:
14842 * bindings/vala/packages/gst-rtsp-server.deps:
14843 * bindings/vala/packages/gst-rtsp-server.files:
14844 * bindings/vala/packages/gst-rtsp-server.gi:
14845 * bindings/vala/packages/gst-rtsp-server.metadata:
14846 * bindings/vala/packages/gst-rtsp-server.namespace:
14847 Added Vala bindings
14849 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14851 * gst/rtsp-server/rtsp-session.c:
14852 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14854 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14856 * examples/Makefile.am:
14857 * gst/rtsp-server/Makefile.am:
14858 Put GStreamer version in library name
14860 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14862 * examples/Makefile.am:
14863 * gst/rtsp-server/Makefile.am:
14864 Fix some issues to pass distcheck
14866 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14868 * gst/rtsp-server/rtsp-server.c:
14869 Added port property to GstRTSPServer class.
14871 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14876 * examples/Makefile.am:
14879 * gst/rtsp-server/Makefile.am:
14880 * gst/rtsp-server/rtsp-client.c:
14881 * gst/rtsp-server/rtsp-client.h:
14882 * gst/rtsp-server/rtsp-media.c:
14883 * gst/rtsp-server/rtsp-media.h:
14884 * gst/rtsp-server/rtsp-server.c:
14885 * gst/rtsp-server/rtsp-server.h:
14886 * gst/rtsp-server/rtsp-session-pool.c:
14887 * gst/rtsp-server/rtsp-session-pool.h:
14888 * gst/rtsp-server/rtsp-session.c:
14889 * gst/rtsp-server/rtsp-session.h:
14891 Split in library and example program
14893 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14895 * src/rtsp-client.h:
14896 Removed obsolete variable
14898 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14900 * src/rtsp-client.c:
14901 * src/rtsp-client.h:
14902 Removed pipeline variable GstRTSPClient, because it's only used in one function
14904 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14906 * src/rtsp-media.c:
14907 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14909 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14911 * src/rtsp-session.c:
14912 Initialize some more vars.
14914 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14916 * src/rtsp-session.c:
14917 Initialize variable to avoid compiler warning.
14919 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14922 Add a reasonable generic .gitignore