1 2022-10-25 09:39:07 +0300 Sebastian Dröge <sebastian@centricular.com>
3 * gst/rtsp-server/rtsp-server.c:
4 Fix various warnings from gobject-introspection
5 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3261>
7 2022-10-15 12:40:04 +0300 Sebastian Dröge <sebastian@centricular.com>
9 * gst/rtsp-server/rtsp-auth.c:
10 * gst/rtsp-server/rtsp-latency-bin.c:
11 * gst/rtsp-server/rtsp-media-factory.c:
12 * gst/rtsp-server/rtsp-media.c:
13 * gst/rtsp-server/rtsp-onvif-media-factory.c:
14 * gst/rtsp-server/rtsp-server.c:
15 * gst/rtsp-server/rtsp-stream.c:
16 rtsp-server: Add/fix various annotations
17 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
19 2022-10-14 08:53:18 +0200 Edward Hervey <edward@centricular.com>
21 * gst/rtsp-server/rtsp-client.h:
22 rtsp-client: Remove duplicate documentation
23 Confuses the documentation builder, since it's documented twice it complains
24 about a missing "Since:" marker whereas it's present in the documentation
26 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
28 2022-08-19 16:16:26 +0200 Linus Svensson <linussn@axis.com>
30 * gst/rtsp-server/rtsp-server.c:
31 rtsp-server: Free client if no connection could be created
32 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>
34 2022-10-11 14:55:48 +0200 Peter Stensson <petest@axis.com>
36 * gst/rtsp-server/rtsp-client.h:
37 rtsp-server: Add since marker for adjust_error_code
38 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>
40 2022-06-21 09:51:55 +0200 Peter Stensson <petest@axis.com>
42 * gst/rtsp-server/rtsp-client.c:
43 * gst/rtsp-server/rtsp-client.h:
44 * gst/rtsp-server/rtsp-media.c:
45 * tests/check/gst/client.c:
46 * tests/check/gst/media.c:
47 rtsp-server: Add support for adjusting request response on pipeline errors
48 The idea is to give the application the possibility to adjust the error
49 code when responding to a request. For that purpose the pipeline's bus
50 messages are emitted to subscribers through a signal handle-message.
51 The subscribers can then check those messages for errors and adjust
52 the response error code by overriding the virtual method
55 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
57 2022-10-04 03:57:31 +0100 Tim-Philipp Müller <tim@centricular.com>
59 * docs/plugins/gst_plugins_cache.json:
62 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
64 === release 1.21.1 ===
66 2022-10-04 01:14:01 +0100 Tim-Philipp Müller <tim@centricular.com>
71 * docs/plugins/gst_plugins_cache.json:
72 * gst-rtsp-server.doap:
76 2022-10-04 01:13:59 +0100 Tim-Philipp Müller <tim@centricular.com>
79 Update ChangeLogs for 1.21.1
81 2022-09-21 19:19:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
84 meson: Use implicit builtin dirs in pkgconfig generation
85 Starting with Meson 0.62, meson automatically populates the variables
86 list in the pkgconfig file if you reference builtin directories in the
87 pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
88 We need this, because ${prefix}/libexec is a hard-coded value which is
89 incorrect on, for example, Debian.
90 Bump requirement to 0.62, and remove version compares that retained
91 support for older Meson versions.
92 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
93 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
95 2021-03-24 14:20:18 -0500 Zebediah Figura <z.figura12@gmail.com>
98 meson: Build with -Wl,-z,nodelete to prevent unloading of dynamic libraries and plugins
99 GLib made the unfortunate decision to prevent libgobject from ever being
100 unloaded, which means that now any library which registers a static type
101 can't ever be unloaded either (and any library that depends on those,
103 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
105 2022-09-05 13:28:18 +1200 Chris Wiggins <chris@chriswiggins.co.nz>
107 * gst/rtsp-server/rtsp-context.c:
108 * gst/rtsp-server/rtsp-context.h:
109 rtsp-server: context: Add method to set the RTSPToken on some RTSPContext
111 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2979>
113 2022-08-24 19:50:19 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
115 * gst/rtsp-server/rtsp-server-internal.h:
116 * gst/rtsp-server/rtsp-stream-transport.c:
117 * gst/rtsp-server/rtsp-stream.c:
118 gst-rtsp-server: Fix pushing backlog to client
119 Check back pressure of a stream transport before popping buffer from its backlog.
120 If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
122 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
124 2022-09-02 16:31:54 +0300 Sebastian Dröge <sebastian@centricular.com>
126 * gst/rtsp-server/rtsp-stream.c:
127 rtsp-server: stream: Don't loop forever if binding to the multicast address fails
128 The address/port is pre-defined by the caller of the function, so
129 retrying is only going to loop forever.
130 Ideally the multicast address should be checked after allocating but
131 this doesn't happen currently, so it's better to error out cleanly then
132 to loop forever trying the same address.
133 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
135 2022-09-01 15:11:31 -0400 Thibault Saunier <tsaunier@igalia.com>
137 * gst/rtsp-sink/meson.build:
139 meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
140 Removing some copy pasted code
141 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
143 2022-09-01 11:51:48 -0400 Thibault Saunier <tsaunier@igalia.com>
146 * gst/rtsp-server/meson.build:
148 meson: Namespace the plugins_doc_dep/libraries variables
149 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
151 2022-08-31 18:44:14 -0400 Thibault Saunier <tsaunier@igalia.com>
154 meson: Rename plugins list and make them "dependency" objects
155 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
157 2022-05-25 18:40:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
159 * gst/rtsp-sink/gstrtspclientsink.c:
160 rtsp+rtmp: Forward warning added to tls-validation-flags to our users
161 With the 2.72 release, glib-networking developers have decided that
162 TLS certificate validation cannot be implemented correctly by them, so
163 they've deprecated it.
164 In a nutshell: a cert can have several validation errors, but there
165 are no guarantees that the TLS backend will return all those errors,
166 and things are made even more complicated by the fact that the list of
167 errors might refer to certs that are added for backwards-compat and
168 won't actually be used by the TLS library.
169 Our best option is to ignore the deprecation and pass the warning onto
170 users so they can make an appropriate security decision regarding
172 We can't deprecate the tls-validation-flags property because it is
173 very useful when connecting to RTSP cameras that will never get
174 updates to fix certificate errors.
175 Relevant upstream merge requests / issues:
176 https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214
177 https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179
178 https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
179 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
181 2022-07-12 16:58:00 +0800 Bruce Liang <Bruce.Liang@Abilitycorp.com.tw>
183 * gst/rtsp-server/rtsp-client.c:
184 rtsp-client: Fix url for generating key in media factory
185 The mount point at / can be accessed by both the URL forms rtsp://<IP>:<PORT> and rtsp://<IP>:<PORT>/.
186 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.
187 This commit creates a new GstRTSPUrl as the normalized url to send to gst_rtsp_media_factory_construct().
188 Fixes:https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1297
189 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2681>
191 2022-06-29 10:55:13 +0100 Tim-Philipp Müller <tim@centricular.com>
194 coding style: allow declarations after statement
195 See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
196 and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
197 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2683>
199 2022-06-14 16:18:35 +0100 Tim-Philipp Müller <tim@centricular.com>
202 * docs/plugins/gst_plugins_cache.json:
203 * docs/plugins/index.md:
204 * docs/plugins/sitemap.txt:
205 docs: make sure rtspclientsink plugin docs index page is called index.html
206 .. instead of plugin-index.html.
207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2592>
209 2022-04-06 12:56:30 +0100 Tim-Philipp Müller <tim@centricular.com>
212 Bump GLib requirement to >= 2.62
213 Can't require 2.64 yet because of
214 https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323
215 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
217 2022-05-16 18:06:16 +0200 Patricia Muscalu <patricia@axis.com>
219 * gst/rtsp-server/rtsp-media.c:
220 rtsp-media: Correct logic on GstRTSPStreamBlocking message reception
221 We must take into account the receiving streams as well when calculating
222 the expected number of the received GstRTSPStreamBlocking messages.
223 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2429>
225 2022-04-27 01:13:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
227 * tests/check/gst/onvif.c:
228 tests/onvif: improve robustness
229 The previous iteration of the code was inferring the type of the
230 frame by looking at the overall size of the gst-payloaded packet.
231 It is more robust to actually parse the payload and look at the
232 actual data buffers it contains.
233 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
235 2022-04-27 01:10:46 +0200 Mathieu Duponchelle <mathieu@centricular.com>
237 * tests/check/gst/onvif.c:
238 tests/onvif: don't push buffers outside segment
239 segment->stop is exclusive, so in reverse playback mode we do not
240 need to output a buffer at that position as it will simply get
242 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
244 2022-02-15 13:39:43 +0000 Pierre Bourré <pierre.moltess@gmail.com>
246 * gst/rtsp-sink/gstrtspclientsink.c:
247 rtspclientsink: fix possible shutdown deadlock collect_streams()
248 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1696>
250 2022-04-07 19:14:27 +0300 Sebastian Dröge <sebastian@centricular.com>
252 * gst/rtsp-server/rtsp-sdp.c:
253 rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME
254 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
256 2022-04-13 14:34:57 +0200 Marc Leeman <m.leeman@televic.com>
258 * gst/rtsp-server/rtsp-stream.c:
259 gst-rtsp-server: minor spelling fixes
260 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>
262 2022-03-25 15:00:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
264 * examples/meson.build:
266 Remove glib and gobject dependencies everywhere
267 They are part of gst_dep already and we have to make sure to always have
268 gst_dep. The order in dependencies matters, because it is also the order
269 in which Meson will set -I args. We want gstreamer's config.h to take
270 precedence over glib's private config.h when it's a subproject.
271 While at it, remove useless fallback args for gmodule/gio dependencies,
272 only gstreamer core needs it.
273 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
275 2022-03-28 21:03:16 +1100 Matthew Waters <matthew@centricular.com>
277 * gst/rtsp-server/rtsp-stream.c:
278 rtsp-stream: remove unused variable:
280 ../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
281 guint n_messages = 0;
283 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
285 2022-03-18 13:42:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
288 meson: Bump all meson requirements to 0.60
289 Lots of new warnings ever since
290 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1934
291 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
293 2022-02-23 17:39:18 +0100 Vivienne Watermeier <vwatermeier@igalia.com>
295 * gst/rtsp-server/rtsp-token.c:
296 documentation: improve misleading wording
297 The documentation for several gst_*_writable_structure functions stated
298 that they would never return NULL, without making clear that the passed
299 object is required to be writable. This changes the wording in those
300 cases to make that requirement more clear.
301 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
303 2022-02-10 08:01:02 +0100 Branko Subasic <branko@axis.com>
305 * examples/test-onvif-server.c:
306 * tests/check/gst/onvif.c:
307 rtponviftimestamp: add support for using reference timestamps
308 Make it posible to configure the element to obtain the timestamps from
309 reference timestamp meta data instead of using the ntp-offset property,
310 or estimating its own offset. Currently the only time format supported
311 is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.
312 In addition the custom event GstNtpOffset has been renamed to
313 GstOnvifTimestamp, to reflect that it is not necessarily used to convey
314 the ntp-offset. As a consequence we had to modify a couple of files in
315 the rtsp-server as well.
317 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
319 2022-02-18 16:05:53 +0100 Branko Subasic <branko@axis.com>
321 * tests/check/gst/onvif.c:
322 * tests/check/gst/rtspserver.c:
323 * tests/check/gst/stream.c:
324 gst-rtsp-server: Plug a few memory leaks in tests
325 Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and
326 gst_stream tests by running the tests in valgrind.
327 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
329 2022-03-07 09:14:46 +0100 Branko Subasic <branko@axis.com>
331 * gst/rtsp-server/rtsp-client.c:
332 gst-rtsp-server: fix race in rtsp-client
333 When tunneling over HTTP, if connection on the second channel happens
334 before the control timer is created we may trigger an assert in
335 rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
336 attaching the client thread to the context.
338 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
340 2022-02-04 11:15:47 +0000 Tim-Philipp Müller <tim@centricular.com>
342 * docs/gst_plugins_cache.json:
345 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
347 === release 1.20.0 ===
349 2022-02-03 19:53:25 +0000 Tim-Philipp Müller <tim@centricular.com>
354 * docs/gst_plugins_cache.json:
355 * gst-rtsp-server.doap:
359 2022-02-03 19:53:18 +0000 Tim-Philipp Müller <tim@centricular.com>
362 Update ChangeLogs for 1.20.0
364 === release 1.19.90 ===
366 2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
371 * docs/gst_plugins_cache.json:
372 * gst-rtsp-server.doap:
376 2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
379 Update ChangeLogs for 1.19.90
381 2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
383 * examples/test-appsrc2.c:
384 gst-rtsp-server: Fix leak in appsrc2 example
385 In the need-data appsrc callback, a buffer is pulled from the
386 appsink. This buffer is then copied so that metadata is writable.
387 The copy is pushed to the appsrc but it doesn't take ownership
388 of the buffer so we need to manually unref it. The original buffer
389 is finally unreffed when the sample is freed.
390 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
392 2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
396 meson: Add explicit check: kwarg to all run_command() calls
397 This is required since Meson 0.61.0, and causes a warning to be
399 https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
400 https://github.com/mesonbuild/meson/issues/9300
401 This exposed a bunch of places where we had broken run_command()
402 calls, unnecessary run_command() calls, and places where check: true
404 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
406 2021-12-20 13:03:34 +0100 Fabrice Fontaine <fontaine.fabrice@gmail.com>
408 * gst/rtsp-server/meson.build:
409 rtsp-server: add gst_dep to gst_rtsp_server_deps
410 Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
411 will avoid the following build failure, because the correct girdir
412 location will be retrieved from gstreamer-1.0.pc:
413 /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
414 Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
415 error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
416 If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
417 Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
419 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
420 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
422 2021-12-16 21:04:53 +0100 Mathieu Duponchelle <mathieu@centricular.com>
424 * gst/rtsp-server/rtsp-stream.c:
425 rtsp-stream: fix get_rates raciness
426 Prior to this patch, we considered that a stream was blocking
427 whenever a pad probe was triggered for either the RTP pad or
429 This led to situations where we subsequently unblocked and expected
430 to find a segment on the RTP pad, which was racy.
431 Instead, we now only consider that the stream is blocking when
432 the pad probe for the RTP pad has triggered with a blockable object
433 (buffer, buffer list, gap event).
434 The RTCP pad is simply blocked without affecting the state of the
437 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
439 2021-11-03 18:44:03 +0000 Tim-Philipp Müller <tim@centricular.com>
441 * docs/gst_plugins_cache.json:
445 === release 1.19.3 ===
447 2021-11-03 15:43:36 +0000 Tim-Philipp Müller <tim@centricular.com>
452 * docs/gst_plugins_cache.json:
453 * gst-rtsp-server.doap:
457 2021-11-03 15:43:32 +0000 Tim-Philipp Müller <tim@centricular.com>
460 Update ChangeLogs for 1.19.3
462 2021-10-25 11:37:45 +0100 Tim-Philipp Müller <tim@centricular.com>
465 meson: require matching GStreamer dep versions for unstable development releases
466 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
467 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>
469 2021-10-18 15:47:00 +0100 Tim-Philipp Müller <tim@centricular.com>
471 * tests/check/meson.build:
472 meson: update for meson.build_root() and .build_source() deprecation
473 -> use meson.project_build_root() or .global_build_root() instead.
474 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
476 2021-10-18 00:40:14 +0100 Tim-Philipp Müller <tim@centricular.com>
479 * tests/check/meson.build:
480 meson: update for dep.get_pkgconfig_variable() deprecation
481 ... in favour of dep.get_variable('foo', ..) which in some
482 cases allows for further cleanups in future since we can
483 extract variables from pkg-config dependencies as well as
484 internal dependencies using this mechanism.
485 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
487 2021-10-01 15:32:58 +0100 Tim-Philipp Müller <tim@centricular.com>
489 * gst/rtsp-server/meson.build:
490 * gst/rtsp-sink/meson.build:
491 rtsp-server: define G_LOG_DOMAIN
493 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
495 2021-10-14 18:38:26 +0100 Tim-Philipp Müller <tim@centricular.com>
498 meson: bump meson requirement to >= 0.59
499 For monorepo build and ugly/bad, for advanced feature
500 option API like get_option('xyz').required(..) which
501 we use in combination with the 'gpl' option.
502 For rest of modules for consistency (people will likely
503 use newer features based on the top-level requirement).
504 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
506 2021-10-12 15:52:48 -0300 Thibault Saunier <tsaunier@igalia.com>
509 meson: Streamline the way we detect when to build documentation
510 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
512 2020-06-27 00:39:00 -0400 Thibault Saunier <tsaunier@igalia.com>
515 * gst/rtsp-server/meson.build:
517 meson: List libraries and their corresponding gir definition
518 Introduces a `libraries` variable that contains all libraries in a
519 list with the following format:
523 'lib': library_object
524 'gir': [ {full gir definition in a dict } ]
529 It therefore refactors the way we build the gir so that we can reuse the
530 same information to build them against 'gstreamer-full' in gst-build
531 when linking statically
532 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
534 2020-06-27 00:37:39 -0400 Thibault Saunier <tsaunier@igalia.com>
536 * gst/rtsp-server/meson.build:
537 meson: Mark files as files()
538 Making it more robust and future proof
539 And fix issues that it creates
540 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
542 2021-10-07 13:00:10 +0300 Sebastian Dröge <sebastian@centricular.com>
544 * gst/rtsp-server/rtsp-media.c:
545 rtsp-media: Unprepare suspended medias too
546 Previously suspended medias immediately reached the UNPREPARED state
547 without going through the media's unprepare() vfunc. This didn't allow
548 the media subclass to do any additional cleanup, and for example the
549 shutdown-eos property of GstRTSPMedia was ignored.
550 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
552 2021-10-06 18:19:29 +0300 Sebastian Dröge <sebastian@centricular.com>
554 * gst/rtsp-server/rtsp-media.c:
555 rtsp-media: Only unprepare a media if it was not already unpreparing anyway
556 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
558 2021-10-03 23:25:23 +0200 Ognyan Tonchev <ognyan@axis.com>
560 * gst/rtsp-server/rtsp-client.c:
561 * gst/rtsp-server/rtsp-session.c:
562 * gst/rtsp-server/rtsp-session.h:
563 rtsp-client: make sure sessmedia will not get freed while used
564 handle_*_request() functions were all retrieving the session media from
565 the session by calling gst_rtsp_session_get_media () which is a transfer-none
566 call. If a session timeout happens at that time, the session media may get freed
567 making the pointer invalid..
569 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
571 2021-10-05 19:37:40 +0300 Sebastian Dröge <sebastian@centricular.com>
573 * gst/rtsp-server/rtsp-media.c:
574 rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
575 Previously the status was only changed for other medias.
576 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
578 2021-10-01 13:51:37 +0300 Sebastian Dröge <sebastian@centricular.com>
580 * gst/rtsp-server/rtsp-session.c:
581 rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
582 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
583 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
585 2021-09-28 10:11:15 +1000 Brad Hards <bradh@frogmouth.net>
588 doc: update IRC links to OFTC
589 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>
591 2021-09-26 01:07:02 +0100 Tim-Philipp Müller <tim@centricular.com>
593 * docs/gst_plugins_cache.json:
596 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
598 === release 1.19.2 ===
600 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
605 * docs/gst_plugins_cache.json:
606 * gst-rtsp-server.doap:
610 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
612 * gst/rtsp-server/rtsp-media.c:
613 * gst/rtsp-server/rtsp-stream.c:
614 * gst/rtsp-server/rtsp-stream.h:
615 * gst/rtsp-sink/gstrtspclientsink.c:
616 Protection against early RTCP packets.
617 When receiving RTCP packets early the funnel is not ready yet and
618 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
619 This causes the thread that handle RTCP packets to go to pause mode.
620 Since this thread is in pause mode there will be no further callbacks to
621 handle keep-alive for incoming RTCP packets. This will make the session
622 time out if the client is not using another keep-alive mechanism.
623 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
624 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
626 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
630 Update COPYING.LIB, COPYING files
631 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
633 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
635 * docs/gst_plugins_cache.json:
639 === release 1.19.1 ===
641 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
646 * docs/gst_plugins_cache.json:
647 * gst-rtsp-server.doap:
651 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
653 * gst/rtsp-server/rtsp-stream.c:
654 rtsp-stream: use new gst_buffer_new_memdup()
655 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
657 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
659 * gst/rtsp-server/rtsp-media-factory-uri.c:
660 rtsp-media: fix leak when adding converter
661 Free the previous caps before reusing the variable for the converter caps.
662 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
664 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
666 * gst/rtsp-server/rtsp-client.c:
667 rtsp-client: fix leak adding headers
668 gst_rtsp_message_add_header() makes a copy of the header, instead
670 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
672 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
674 * gst/rtsp-server/rtsp-stream.c:
675 Use gst_element_request_pad_simple...
676 Instead of the deprecated gst_element_get_request_pad.
677 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
679 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
681 * gst/rtsp-server/rtsp-media.c:
682 rtsp-media: Ensure the bus watch is removed during unprepare
683 It's possible for the destruction of the source to be delayed.
684 Instead of relying on the dispose() to remove the bus watch, do
686 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
688 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
691 docs: minor spelling correction in README
692 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
694 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
696 * examples/test-replay-server.c:
697 test-replay-server: minor spelling corrections
698 Bumped on these while investigating the example code.
699 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
701 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
703 * tests/check/gst/stream.c:
704 tests: Don't fail tests if IPv6 not available.
705 On computers with IPv6 disabled it shouldn't result in a test failure.
706 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
708 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
710 * gst/rtsp-server/rtsp-media.c:
711 rtsp-media: Add one more case to seek avoidance
712 This is an extension to the previous commit. There can also be cases where the
713 start position is not specified, in those cases we should also avoid doing
714 seeking unless it's forced.
715 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
717 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
719 * gst/rtsp-server/rtsp-media.c:
720 rtsp-media: Improve skipping trickmode seek.
721 We can also skip the seek if the end range is already
723 Avoids initial seek on play start if playing full stream.
724 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
726 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
728 * gst/rtsp-sink/gstrtspclientsink.c:
729 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
730 It's sufficient to run them during the FIRST stage instead of in both.
731 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
733 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
735 * tests/check/gst/rtspclientsink.c:
736 tests: rtspclientsink: fix some leaks
737 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
739 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
741 * gst/rtsp-sink/gstrtspclientsink.c:
742 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
743 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
745 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
747 * tests/check/gst/rtspclientsink.c:
748 rtspclientsink: add unit test for potential shutdown deadlock
749 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
751 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
753 * gst/rtsp-sink/gstrtspclientsink.c:
754 rtspclientsink: fix deadlock on shutdown before preroll
755 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
756 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
758 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
760 * gst/rtsp-server/rtsp-stream.c:
761 rtsp-stream: avoid deadlock in send_func
762 Currently the send_func() runs in a thread of its own which is started
763 the first time we enter handle_new_sample(). It runs in an outer loop
764 until priv->continue_sending is FALSE, which happens when a TEARDOWN
765 request is received. We use a local variable, cont, which is initialized
766 to TRUE, meaning that we will always enter the outer loop, and at the
767 end of the outer loop we assign it the value of priv->continue_sending.
768 Within the outer loop there is an inner loop, where we wait to be
769 signaled when there is more data to send. The inner loop is exited when
770 priv->send_cookie has changed value, which it does when more data is
771 available or when a TEARDOWN has been received.
772 But if we get a TEARDOWN before send_func() is entered we will get stuck
773 in the inner loop because no one will increase priv->session_cookie
775 By not entering the outer loop in send_func() if priv->continue_sending
776 is FALSE we make sure that we do not get stuck in send_func()'s inner
777 loop should we receive a TEARDOWN before the send thread has started.
778 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
779 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
781 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
783 * gst/rtsp-server/rtsp-client.c:
784 rtsp-client: cleanup transports during TEARDOWN
785 When tunneling RTP over RTSP the stream transports are stored in a hash
786 table in the GstRTSPClientPrivate struct. They are used for, among other
787 things, mapping channel id to stream transports when receiving data from
788 the client. The stream tranports are created and added to the hash table
789 in handle_setup_request(), but unfortuately they are not removed in
790 handle_teardown_request(). This means that if the client sends data on
791 the RTSP connection after it has sent the TEARDOWN, which is often the
792 case when audio backchannel is enabled, handle_data() will still be able
793 to map the channel to a session transport and pass the data along to it.
794 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
795 because the stream is no longer joined to a bin.
796 We avoid this by removing the stream transports from the hash table when
797 we handle the TEARDOWN request.
798 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
800 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
802 * docs/gst_plugins_cache.json:
803 * gst/rtsp-sink/gstrtspclientsink.c:
804 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
805 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
807 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
809 * tests/check/gst/client.c:
810 Add test cases for mountpoint of '/'
811 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
813 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
815 * gst/rtsp-server/rtsp-client.c:
816 * gst/rtsp-server/rtsp-mount-points.c:
817 * gst/rtsp-server/rtsp-session-media.c:
818 Make a mount point of "/" work correctly.
819 As far as I can tell, this is neither explicitly allowed nor
820 forbidden by RFC 7826.
821 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
822 use in the wild (presumably with non-GStreamer servers).
823 GStreamer's prior behavior was confusing, in that
824 gst_rtsp_mount_points_add_factory() would appear to accept a mount
825 path of "" or "/", but later connection attempts would fail with a
826 "media not found" error.
827 This commit makes a mount path of "/" work for either form of URL,
828 while an empty mount path ("") is rejected and logs a warning.
829 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
831 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
833 * docs/gst_plugins_cache.json:
834 * gst/rtsp-sink/gstrtspclientsink.c:
835 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
836 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
838 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
840 * gst/rtsp-server/rtsp-media.c:
841 rtsp-media: Only count senders when counting blocked streams
842 Only sender streams sends the GstRTSPStreamBlocking message, so only
843 these should be counted before setting media status to prepared.
844 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
846 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
848 * gst/rtsp-sink/gstrtspclientsink.c:
849 rtspclientsink add proper support for uri queries
850 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
852 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
854 * gst/rtsp-server/rtsp-client.c:
855 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
856 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
857 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
859 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
861 * gst/rtsp-server/rtsp-stream.c:
862 rtsp-stream: collect a clock_rate when blocking
863 This lets us provide a clock_rate in a fashion similar to the
864 other code paths in get_rtpinfo()
865 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
867 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
869 * gst/rtsp-server/rtsp-media.c:
870 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
871 Otherwise this will cause memory corruption as the property expects a 64
873 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
875 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
877 * gst/rtsp-server/rtsp-media.c:
878 * gst/rtsp-server/rtsp-stream.c:
879 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
880 To prevent cases with prerolling when the inactive stream prerolls first
881 and the server proceeds without waiting for the active stream, we will
882 ignore GstRTSPStreamBlocking messages from incomplete streams. When
883 there are no complete streams (during DESCRIBE), we will listen to all
885 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
887 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
889 * tests/check/gst/media.c:
890 * tests/check/meson.build:
891 * tests/files/test.avi:
892 media test: Add test for seeking one active stream with a demuxer
893 Add another seek_one_active_stream test but with a demuxer. The demuxer
894 will flush both streams in opposed to the existing test which only
895 flushes the active stream. This will help exposing problems with the
896 prerolling process after a flushing seek.
897 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
899 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
901 * gst/rtsp-server/meson.build:
903 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
904 * pkgconfig/gstreamer-rtsp-server.pc.in:
905 * pkgconfig/meson.build:
906 Meson: Use pkg-config generator
907 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
909 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
912 meson: update glib minimum version to 2.56
913 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
915 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
917 * examples/test-launch.c:
918 * gst/rtsp-server/rtsp-media-factory.c:
919 * gst/rtsp-server/rtsp-media-factory.h:
920 * gst/rtsp-server/rtsp-media.c:
921 * gst/rtsp-server/rtsp-server-internal.h:
922 * gst/rtsp-server/rtsp-stream.c:
923 * tests/check/gst/client.c:
924 rtsp-media-factory: expose API to disable RTCP
925 This is supported by the RFC, and can be useful on systems where
926 allocating two consecutive ports is problematic, and RTCP is not
928 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
930 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
932 * hooks/pre-commit.hook:
934 git: use our standard pre commit hook
935 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
937 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
939 * gst/rtsp-server/rtsp-stream.c:
940 rtsp-stream: make use of blocked_running_time in query_position
941 When blocking, the sink element will not have received a buffer
942 yet and the position query will fail. Instead, we make use of
943 the running time of the buffer we blocked on.
944 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
946 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
948 * gst/rtsp-server/rtsp-stream.c:
949 rtsp-stream: collect rtp info when blocking
950 We don't unblock the stream anymore before replying to the
951 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
952 so the sinks don't have a last-sample after potentially flush
953 seeking. seek_trickmode waits for preroll however, which means
954 the stream will block and wait for a first buffer. Subsequent
955 calls to get_rtpinfo() can thus make use of the information.
956 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
957 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
959 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
961 * examples/meson.build:
962 * examples/test-replay-server.c:
963 * examples/test-replay-server.h:
964 examples: Add an example for loop playback
965 This demo example shows a way of file loop playback of a given source.
966 Note that client seek request is not properly implemented yet.
967 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
969 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
971 * gst/rtsp-server/rtsp-media.c:
972 rtsp-media: Plug memory leak
973 The get-storage signal of rtpbin increases the ref count of the storage.
974 So we have to unref it after usage.
975 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
977 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
979 * gst/rtsp-server/rtsp-media.c:
980 rtsp-media: Get rates only on sender streams
981 When play a media with both sender and receiver stream, like ONVIF
982 back channel audio in, gst_rtsp_media_get_rates call
983 gst_rtsp_stream_get_rates for each stream to set the rates. But
984 gst_rtsp_stream_get_rates return false for the receiver steam, which
985 lead a g_assert crash.
986 Instead to get rates on all streams, now just get rates on sender
988 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
990 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
992 * gst/rtsp-server/rtsp-media.c:
993 * gst/rtsp-server/rtsp-server-internal.h:
994 * gst/rtsp-server/rtsp-stream.c:
995 rtsp-media: set a 0 storage size for TCP receivers
996 ulpfec correction is obviously useless when receiving a stream
997 over TCP, and in TCP modes the rtp storage receives non
998 timestamped buffers, causing it to queue buffers indefinitely,
999 until the queue grows so large that sanity checks kick in and
1000 warnings start to get emitted.
1001 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
1003 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1005 * gst/rtsp-server/rtsp-stream.c:
1006 rtsp-stream: preroll on gap events
1007 This allows negotiating a SDP with all streams present, but only
1008 start sending packets at some later point in time
1009 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
1011 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1013 * gst/rtsp-server/rtsp-media.c:
1014 rtsp-media: do not unblock on unsuspend
1015 rtsp_media_unsuspend() is called from handle_play_request()
1016 before sending the play response. Unblocking the streams here
1017 was causing data to be sent out before the client was ready
1018 to handle it, with obvious side effects such as initial packets
1019 getting discarded, causing decoding errors.
1020 Instead we can simply let the media streams be unblocked when
1021 the state of the media is set to PLAYING, which occurs after
1022 sending the play response.
1023 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
1025 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1028 ci: include template from gst-ci master branch again
1030 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
1032 * docs/gst_plugins_cache.json:
1036 === release 1.18.0 ===
1038 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1044 * docs/gst_plugins_cache.json:
1045 * gst-rtsp-server.doap:
1049 === release 1.17.90 ===
1051 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1056 * docs/gst_plugins_cache.json:
1057 * gst-rtsp-server.doap:
1061 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
1063 * gst/rtsp-server/rtsp-thread-pool.c:
1064 rtsp-thread-pool.c: fix clang 10 warning
1065 clang 10 is complaining about incompatible types due to the
1068 ../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]
1070 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1072 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
1074 * gst/rtsp-server/rtsp-thread-pool.c:
1075 rtsp-thread-pool.c: fix clang 10 warning
1076 clang 10 is complaining about incompatible types due to the
1079 ../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]
1081 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
1083 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
1085 * gst/rtsp-server/rtsp-sdp.c:
1086 rtsp-sdp: Fix resource leak in mikey messsage
1087 Fixed a resource leak for mikey message while adding crypto session
1089 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
1091 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
1094 * scripts/extract-release-date-from-doap-file.py:
1095 meson: set release date from .doap file for releases
1096 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
1098 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1100 * gst/rtsp-server/rtsp-stream.c:
1101 rtsp-stream: explicitly set caps on udpsrc elements
1102 This causes them to send caps events before data flow, which is
1103 usually a pretty correct thing to do!
1104 Not doing so manifested in a bug where ssrcdemux wouldn't forward
1105 the caps it had received with an extra ssrc field, as it hadn't
1106 received any caps event.
1108 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
1110 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1112 * docs/gst_plugins_cache.json:
1116 === release 1.17.2 ===
1118 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1123 * docs/gst_plugins_cache.json:
1124 * gst-rtsp-server.doap:
1128 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
1130 * docs/gst_plugins_cache.json:
1131 doc: Stop documenting properties from parents
1133 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1135 * docs/gst_plugins_cache.json:
1136 docs: Fix version in the plugins cache
1137 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1139 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1141 * gst/rtsp-sink/gstrtspclientsink.c:
1142 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
1143 It's deprecated, unneeded and doesn't do anything anymore.
1144 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
1146 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
1151 === release 1.17.1 ===
1153 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
1158 * docs/gst_plugins_cache.json:
1159 * gst-rtsp-server.doap:
1163 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1165 * gst/rtsp-server/rtsp-media.c:
1166 rtsp-media: Add/configure transports when completing the pipeline
1167 Otherwise the transports are not set up yet during the PLAY request
1168 handling when unsuspending (and thus unblocking) the media.
1169 In case of live pipelines this then causes the first few packets to go
1170 to the sinks before they know what to do with them, and they simply
1171 discard them which is rather suboptimal in case of keyframes.
1172 For non-live pipelines this is not a problem because the sink will still
1173 be PAUSED and as such not send out the data yet but wait until it goes
1174 to PLAYING, which is late enough.
1175 Adding the transports multiple times is not a problem: if the transport
1176 is already added it won't be added another time and TRUE will be
1178 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
1180 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
1181 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1183 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1185 * gst/rtsp-server/rtsp-media.c:
1186 rtsp-media: Fix misleading comment
1187 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1189 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
1191 * gst/rtsp-server/rtsp-media.c:
1192 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
1193 The pad probes are not needed anymore at this point and later when
1194 reaching buffering 100% only the state is changed, no unblocking
1196 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1198 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
1200 * gst/rtsp-server/rtsp-media.c:
1201 rtsp-media: Remove duplicated media_unblock() function
1202 It does literally the same as media_streams_set_blocked(FALSE).
1203 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
1205 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
1207 * examples/test-onvif-server.c:
1208 test-onvif-server: cast ntp-offset property value to 64 bit
1209 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
1211 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
1213 * docs/gst_plugins_cache.json:
1214 docs: Update plugins cache
1216 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1218 * examples/test-onvif-server.c:
1219 * examples/test-onvif-server.h:
1220 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1221 onvif-media-factory: define autoptr cleanup function
1222 And have the factory in the onvif-server example inherit from
1223 GstRTSPOnvifMediaFactory.
1224 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
1226 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
1228 * docs/gst_plugins_cache.json:
1229 docs: Update plugins cache
1231 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
1233 * tests/check/gst/rtspserver.c:
1234 tests: enforce I420 format
1235 Test was not enforcing a video format on videotestsrc. I420 was picked as it
1236 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
1237 true (gst-plugins-base!689).
1238 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
1240 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1242 * gst/rtsp-sink/gstrtspclientsink.c:
1243 plugins: uddate gst_type_mark_as_plugin_api() calls
1245 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
1248 doc: Require hotdoc >= 0.11.0
1250 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1252 * docs/gst_plugins_cache.json:
1253 docs: Update gst_plugins_cache.json
1255 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1257 * gst/rtsp-sink/gstrtspclientsink.c:
1258 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
1260 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
1262 * gst/rtsp-server/meson.build:
1263 meson: gir: remove bogus sources_top_dir kwarg
1264 Doesn't actually exist. Was fixed differently in Meson
1265 so that the user doesn't have to specify it.
1266 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
1268 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1270 * tests/check/meson.build:
1271 tests: put registry into tests/check not the gst/ subdir
1272 Underscorify the test name before setting GST_REGISTRY,
1273 so the registry actually ends up in the current build dir
1274 and not some subdir.
1275 For consistency with the other modules, but should also
1276 avoid problems on windows.
1277 Also fix indentation of environment block.
1278 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1280 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
1282 * tests/check/meson.build:
1283 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
1284 If core is built as a subproject (e.g. as in gst-build), make sure to use
1285 the gst-plugin-scanner from the built subproject. Without this, gstreamer
1286 might accidentally use the gst-plugin-scanner from the install prefix if
1287 that exists, which in turn might drag in gst library versions we didn't
1288 mean to drag in. Those gst library versions might then be older than
1289 what our current build needs, and might cause our newly-built plugins
1290 to get blacklisted in the test registry because they rely on a symbol
1291 that the wrongly-pulled in gst lib doesn't have.
1292 This should fix running of unit tests in gst-build when invoking
1293 meson test or ninja test from outside the devenv for the case where
1294 there is an older or different-version gst-plugin-scanner installed
1295 in the install prefix.
1296 In case no gst-plugin-scanner is installed in the install prefix, this
1297 will fix "GStreamer-WARNING: External plugin loader failed. This most
1298 likely means that the plugin loader helper binary was not found or
1299 could not be run. You might need to set the GST_PLUGIN_SCANNER
1300 environment variable if your setup is unusual." warnings when running
1302 In the case where we find GStreamer core via pkg-config we use
1303 a newly-added pkg-config var "pluginscannerdir" to get the right
1304 directory. This has the benefit of working transparently for both
1305 installed and uninstalled pkg-config files/setups.
1306 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1308 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
1310 * tests/check/meson.build:
1311 tests: gst-plugins-base and -bad plugins are required for the unit tests
1312 Make hard requirement until we have more fine-grained control
1313 in the unit tests. Of course the presence of the .pc file doesn't
1314 imply that the plugins we need are actually there, but it's at
1315 least a step in the right direction.
1316 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1318 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1320 * tests/check/meson.build:
1321 tests: pick up rtsp-server plugins from build directory only
1322 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
1324 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
1326 * gst/rtsp-server/rtsp-media.c:
1327 rtsp-media: wait for all GstRTSPStreamBlocking messages
1328 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
1329 each active stream when checking if all streams are blocked.
1330 Without this change there will be a race condition when using two or
1331 more streams and rtsp-media receives a GstRTSPStreamBlocking message
1332 from one of the streams. This is because rtsp-media then checks if all
1333 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
1334 stream. This function call returns TRUE if the stream has sent a
1335 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
1336 receive this message. This would then result in that rtsp-media
1337 erroneously thinks it is blocking all streams which could result in
1338 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
1339 preroll, this could result in that rtsp-media thinks that the pipeline
1340 is prerolled even though that might not be the case.
1341 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
1343 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
1345 * gst/rtsp-server/rtsp-media.c:
1346 rtsp-media: update expected_async_done during suspend
1347 Set expected_async_done to FALSE in default_suspend() if a state change
1348 occurs and the return value from set_target_state() is something other
1349 than GST_STATE_CHANGE_ASYNC.
1350 Without this change there is a risk that expected_async_done will be
1351 TRUE even though no asynchronous state change is taking place. This
1352 could happen if the pipeline is set to PAUSED using
1353 media_set_pipeline_state_locked(), an asynchronous state change starts
1354 and then the media is suspended (which could result in a state change,
1355 aborting the asynchronous state change). If the media is suspended
1356 before the asynchronous state change ends then expected_async_done will
1357 be TRUE but no asynchronous state change is taking place.
1358 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
1360 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
1362 * gst/rtsp-server/rtsp-client.c:
1363 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
1364 There was a race condition where client was being finalized and
1365 concurrently in some other thread the rtsp ctrl timout was relying on
1366 client data that was being freed.
1367 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
1368 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
1370 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1372 * gst/rtsp-server/rtsp-media-factory.c:
1373 * gst/rtsp-server/rtsp-media-factory.h:
1374 * gst/rtsp-server/rtsp-media.c:
1375 * gst/rtsp-server/rtsp-media.h:
1376 media-factory: complete DSCP QoS setting support
1377 add dscp_qos setting support at factory and media level to setup IP DSCP
1378 field of bounded UDP sinks.
1379 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
1380 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
1382 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1384 * gst/rtsp-server/rtsp-client.c:
1385 rtsp-client: Fix some race conditions around timeout source removal
1386 We always need to take the lock while accessing it as otherwise another
1387 thread might've removed it in the meantime. Also when destroying and
1388 creating a new one, ensure that the mutex is not shortly unlocked in
1389 between as during that time another one might potentially be created
1391 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
1393 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
1395 * gst/rtsp-server/rtsp-media.c:
1396 * gst/rtsp-server/rtsp-stream.c:
1397 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
1398 And the same for gst_rtsp_stream_get_rates().
1399 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
1401 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1403 * examples/test-onvif-server.c:
1404 examples: test-onvif-server: fix compiler warnings on raspbian
1405 Fix printf format for 64-bit variables.
1406 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
1408 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
1410 * gst/rtsp-server/rtsp-stream-transport.c:
1411 * gst/rtsp-server/rtsp-stream-transport.h:
1412 * gst/rtsp-server/rtsp-stream.c:
1413 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
1414 The old API is preserved now and new API was added that provides the
1415 additional parameter to the callback.
1416 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
1417 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
1419 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
1421 * gst/rtsp-server/rtsp-client.c:
1422 rtsp-client: Store the timeout source by pointer instead of id
1423 That way we don't have to retrieve it again from the main context when
1424 destroying it but can directly do so.
1425 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1427 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
1429 * gst/rtsp-server/rtsp-client.c:
1430 rtsp-client: Clean up watch/watch context and related state consistently
1431 And assert that it was cleaned up properly before the client is
1432 finalized. If something is still around when the client is shut down
1433 then something went very wrong before.
1434 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1436 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1438 * gst/rtsp-server/rtsp-client.c:
1439 * tests/check/gst/rtspserver.c:
1440 rtsp-client: Combine the pre-session and post-session timeout
1441 They previously used the same state but different mechanisms and
1442 functions, which was difficult to follow, error prone and simply
1444 Also adjust the test for the post-session timeout a bit to be less racy
1445 now that the timing has slightly changed.
1446 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1448 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1450 * gst/rtsp-server/rtsp-client.c:
1451 rtsp-client: Don't ever close the client connection directly when a session is torn down
1452 There might be other sessions that are running over the same RTSP
1453 connection and we should not simply close the client directly if one of
1455 By default the connection will be closed once the client closes it or
1456 the OS does. This behaviour can be adjusted with the
1457 post-session-timeout property, which allows to close it automatically
1458 from the server side after all sessions are gone and the given timeout
1460 This reverts the previous commit.
1461 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
1463 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
1465 * gst/rtsp-server/rtsp-client.c:
1466 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
1467 Instead of closing it never at all. Previously there was only code that
1468 closed the client asynchronously if sending the response happened
1469 asynchrously at a later time.
1470 Thanks to Christian M for debugging this issue.
1471 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
1472 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
1474 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
1476 * gst/rtsp-server/rtsp-stream.c:
1477 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
1478 Otherwise no sink is found for multicast sreams and the less accurate
1479 fallback is used to determine the current sequence number and timestamp.
1481 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1483 * gst/rtsp-server/rtsp-auth.c:
1484 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
1485 When using the basic authentication scheme, we wouldn't validate that
1486 the authorization field of the credentials is not NULL and pass it on
1487 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
1488 dereference the NULL pointer and crash.
1489 A specially crafted (read: invalid) RTSP header can cause this to
1491 As a solution, check for the authorization to be not NULL before
1492 continuing processing it and if it is simply fail authentication.
1493 This fixes CVE-2020-6095 and TALOS-2020-1018.
1494 Discovered by Peter Wang of Cisco ASIG.
1496 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
1498 * gst/rtsp-server/rtsp-client.c:
1499 rtsp-client: Use watch_context before unref
1500 Move the usage of priv->watch_context to beginning of function
1501 gst_rtsp_client_finalize. Instead of use it after
1502 g_main_context_unref (priv->watch_context).
1504 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1506 * gst/rtsp-server/rtsp-stream.c:
1507 rtsp-stream: fix deadlock on transport removal
1508 We cannot take the RTSPStream lock while holding a transport backlog
1509 lock, as remove_transport may be called externally, which will
1510 take first the RTSPStream lock then the transport backlog lock.
1512 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1514 * gst/rtsp-server/rtsp-server-internal.h:
1515 * gst/rtsp-server/rtsp-stream-transport.c:
1516 * gst/rtsp-server/rtsp-stream.c:
1517 rtsp-stream: clear backlog when removing transport
1518 This ensures we don't end up calling any of transports' callbacks
1519 with a potentially unreffed user_data (in practice, a client that
1520 may have been removed)
1522 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1524 * gst/rtsp-server/rtsp-stream.c:
1525 rtsp-stream: marshal calls to send_tcp_message to a single thread
1526 In order to address the race condition pointed out at
1527 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
1528 we get rid of the send thread pool, and instead spawn and manage
1529 a single thread to pull samples from app sinks and add them to
1530 the transport's backlogs.
1531 Additionally, we now also always go through the backlogs in order
1532 to simplify the logic.
1534 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1536 * gst/rtsp-server/rtsp-server-internal.h:
1537 * gst/rtsp-server/rtsp-stream-transport.c:
1538 * gst/rtsp-server/rtsp-stream.c:
1539 rtsp-stream: properly protect TCP backlog access
1541 We cannot hold stream->lock while pushing data, but need
1542 to consistently check the state of the backlog both from
1543 the send_tcp_message function and the on_message_sent function,
1544 which may or may not be called from the same thread.
1545 This commit introduces internal API to allow for potentially
1546 recursive locking of transport streams, addressing a race
1547 condition where the RTSP stream could push items out of order
1548 when popping them from the backlog.
1550 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1552 * gst/rtsp-server/rtsp-media.c:
1553 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
1554 It's taken ownership of by the media, and returned with `transfer none`
1555 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
1556 first then any bindings will wrongly take ownership of the pipeline once
1557 it arrives in bindings code.
1559 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
1561 * examples/test-onvif-client.c:
1562 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
1564 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
1566 * gst/rtsp-server/rtsp-media.c:
1567 rtsp-media: fix default latency
1569 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1571 * gst/rtsp-server/rtsp-client.c:
1572 rtsp-client: make closing more thread safe
1573 + Take the watch lock prior to using priv->watch
1574 + Flush both the watch and connection before closing / unreffing
1575 gst_rtsp_connection_close() is not threadsafe on its own, this is
1576 a workaround at the client level, where we control both the watch
1579 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
1581 * gst/rtsp-server/rtsp-latency-bin.c:
1582 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
1585 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
1586 `your_type_get_instance_private()` function instead
1589 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
1591 * gst/rtsp-server/rtsp-client.c:
1592 * tests/check/gst/rtspserver.c:
1593 rtsp-client: add property post-session-timeout
1594 This is a TCP connection timeout for client connections, in seconds.
1595 If a positive value is set for this property, the client connection
1596 will be kept alive for this amount of seconds after the last session
1597 timeout. For negative values of this property the connection timeout
1598 handling is delegated to the system (just as it was before).
1601 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1603 * gst/rtsp-server/rtsp-stream.c:
1604 rtsp-stream: check for NULL transports prior to ref'ing
1606 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1608 * gst/rtsp-server/rtsp-server-internal.h:
1609 * gst/rtsp-server/rtsp-stream-transport.c:
1610 * gst/rtsp-server/rtsp-stream.c:
1611 rtsp-stream: fix checking of TCP backpressure
1612 The internal index of our appsinks, while it can be used to
1613 determine whether a message is RTP or RTCP, is not necessarily
1614 the same as the interleaved channel. Let the stream-transport
1615 determine the channel to check backpressure for, the same way
1616 it determines the channel according to whether it is sending
1619 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1621 * gst/rtsp-server/rtsp-session.c:
1622 rtsp-session: Butcher the file to please gst-indent in the CI
1623 This should be reverted once the CI has an updated gst-indent.
1625 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1627 * gst/rtsp-server/rtsp-session.c:
1628 * gst/rtsp-server/rtsp-session.h:
1629 * gst/rtsp-sink/gstrtspclientsink.c:
1630 * gst/rtsp-sink/gstrtspclientsink.h:
1631 rtsp-session & client: Remove deprecated GTimeVal
1632 GTimeVal won't work past 2038
1634 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1636 * gst/rtsp-server/rtsp-auth.c:
1637 rtsp-auth: fix default token leak
1639 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1641 * gst/rtsp-sink/gstrtspclientsink.c:
1642 gstrtspclientsink: unref transports when closing bin
1645 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1647 * gst/rtsp-server/rtsp-media.c:
1648 rtsp-media: Force seek when flush flag is set
1649 The commit "rtsp-client: define all seek accuracy flags from
1650 setup_play_mode" changed the behaviour of when doing a seek.
1651 Before that commit, having the flush flag set would result in a seek
1653 Even if no seek was needed. One reason to force seek is to flush old buffers
1654 created in Describe requests.
1655 Thus adding force seek also for flush flag will result in play request
1658 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1660 * gst/rtsp-server/rtsp-client.c:
1661 rtsp-client: Revitalize dead code
1662 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1665 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1667 * gst/rtsp-server/rtsp-sdp.c:
1668 rtsp-sdp: Don't try to use non-initialized values
1669 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1670 returns TRUE. Also avoid the whole clock signalling block if we're not
1671 dealing with senders.
1676 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1678 * gst/rtsp-server/rtsp-stream-transport.c:
1679 * gst/rtsp-server/rtsp-stream.c:
1680 * tests/check/gst/stream.c:
1681 rtsp-stream: Removing invalid transports returns false
1682 When removing transports an assertion was that the transports passed in
1683 for removal are present in the list, however that can't be assumed.
1684 As an example if a transport was removed from a thread running
1685 send_tcp_message, the main thread can try to remove the same transport
1686 again if it gets a handle_pause_request. This will not effect the
1687 transport list but it will effect n_tcp_transports as it will be
1688 decrement and then have the wrong value.
1690 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1692 * tests/check/gst/client.c:
1693 client test: add scale and speed negative tests
1694 Negative tests for scale and speed should be done as well, verify that
1695 the response code is "400 Bad request" when a bad request is done.
1697 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1699 * gst/rtsp-server/rtsp-auth.c:
1700 * gst/rtsp-server/rtsp-client.c:
1701 * gst/rtsp-server/rtsp-media-factory.c:
1702 * gst/rtsp-server/rtsp-media.c:
1703 * gst/rtsp-server/rtsp-server.c:
1704 * gst/rtsp-server/rtsp-session-pool.c:
1705 * gst/rtsp-server/rtsp-stream.c:
1706 * gst/rtsp-sink/gstrtspclientsink.c:
1707 Don't pass default GLib marshallers for signals
1708 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1709 actually internally optimize the signal (if the marshaller is available
1710 in GLib itself) by also setting the valist marshaller. This makes the
1711 signal emission a bit more performant than the regular marshalling,
1712 which still needs to box into `GValue` and call libffi in case of a
1714 Note that for custom marshallers, one would use
1715 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1717 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1719 * gst/rtsp-server/rtsp-mount-points.c:
1720 GstRTSPMountPoints: Remove any existing factory before adding a new one
1721 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1722 previous mount point will be freed" which was true when it was
1723 implemented using a GHashTable. But in 2012 it got rewrote using a
1724 GSequence and since then it could have 2 factories for the same path.
1725 Which one gets used is random, depending on the sorting order of 2
1728 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1730 * gst/rtsp-server/rtsp-client.c:
1731 * gst/rtsp-server/rtsp-server-internal.h:
1732 * gst/rtsp-server/rtsp-stream-transport.c:
1733 * gst/rtsp-server/rtsp-stream-transport.h:
1734 * gst/rtsp-server/rtsp-stream.c:
1735 stream: refactor TCP backpressure handling
1736 The previous implementation stopped sending TCP messages to
1737 all clients when a single one stopped consuming them, which
1738 obviously created problems for shared media.
1739 Instead, we now manage a backlog in stream-transport, and slow
1740 clients are removed once this backlog exceeds a maximum duration,
1741 currently hardcoded.
1744 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1747 meson: build gir even when cross-compiling if introspection was enabled explicitly
1748 This can be made to work in certain circumstances when
1749 cross-compiling, so default to not building g-i stuff
1750 when cross-compiling, but allow it if introspection was
1751 enabled explicitly via -Dintrospection=enabled.
1752 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1754 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1756 * gst/rtsp-server/rtsp-session.c:
1757 rtsp-session: clean up comment extra-timeout
1759 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1761 * gst/rtsp-server/rtsp-client.c:
1762 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1763 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1764 from the RTSP context.
1767 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1769 * gst/rtsp-server/rtsp-client.c:
1770 * gst/rtsp-server/rtsp-media.c:
1771 * gst/rtsp-server/rtsp-media.h:
1772 rtsp-client: Lock shared media
1773 For shared media we got race conditions. Concurrently rtsp clients might
1774 suspend or unsuspend the shared media and thus change the state without
1775 the clients expecting that.
1776 By introducing a lock that can be taken by callers such as rtsp_client
1777 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1778 to handle the media sequentially thus allowing one client to finish its
1779 rtsp call before another client calls on the same media.
1780 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1783 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1785 * gst/rtsp-server/rtsp-session.c:
1786 rtsp-session: add property extra-timeout
1787 Extra time to add to the timeout, in seconds. This only
1788 affects the time until a session is considered timed out
1789 and is not signalled in the RTSP request responses.
1790 Only the value of the timeout property is signalled in the
1793 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1795 * gst/rtsp-server/rtsp-stream.c:
1796 rtsp-stream : fix race condition in send_tcp_message
1797 If one thread is inside the send_tcp_message function and are done
1798 sending rtp or rtcp messages so the n_outstanding variable is zero
1799 however have not exit the loop sending the messages. While sending its
1800 messages, transports have been added or removed to the transport list,
1801 so the cache should be updated. If now an additional thread comes to
1802 the function send_tcp_message and trying to send rtp messages it will
1803 first destroy the rtp cache that is still being iterated trough by the
1807 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1816 * examples/.gitignore:
1817 * examples/Makefile.am:
1819 * gst/rtsp-server/.gitignore:
1820 * gst/rtsp-server/Makefile.am:
1821 * gst/rtsp-sink/Makefile.am:
1822 * pkgconfig/.gitignore:
1823 * pkgconfig/Makefile.am:
1825 * tests/Makefile.am:
1826 * tests/check/Makefile.am:
1827 Remove autotools build
1829 Maybe we can now use the meson pkgconfig module
1830 for .pc files? (Does it support uninstalled now?)
1832 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1834 * tests/check/gst/client.c:
1835 client: fix test mem leak in attach_rate_tweaking_probe
1837 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1839 * tests/check/gst/media.c:
1840 media: remove memleak in test test_media_seek
1842 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1844 * tests/check/gst/rtspserver.c:
1845 rtspserver: Remove memleak in test test_double_play
1847 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1849 * gst/rtsp-server/rtsp-media.c:
1850 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1852 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1854 * gst/rtsp-server/rtsp-media.c:
1855 * tests/check/gst/rtspserver.c:
1856 rtsp-media: Unblock all streams
1857 When unsuspending and going to PLAYING, unblock all streams instead of
1858 only those that are linked (the linked streams are the ones for which
1859 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1860 pushing buffers on unlinked streams.
1861 This change is because playback using single-threaded demuxers like
1862 matroska-demux could be blocked if SETUP was not called for all media.
1863 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1864 gstflvdemux, qtdemux, and matroska-demux) will handle
1865 GST_FLOW_NOT_LINKED automatically.
1868 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1870 * gst/rtsp-server/rtsp-media.c:
1871 * tests/check/gst/rtspserver.c:
1872 rtsp-media: Wait on async when needed.
1873 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1874 In the unit test the pause from adjust_play_mode will cause a preroll
1875 and after that async-done will be produced.
1876 Without this patch there are no one consuming this async-done and when
1877 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1878 wait for async-done. But then it wrongly find the async-done prodused by
1879 adjus_play_mode and continue executing without waiting for the preroll
1882 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1884 * gst/rtsp-server/rtsp-client.c:
1885 rtsp-client: RTP Info when completed_sender
1886 Change condition that should be fulfilled regarding RTPInfo.
1887 Replace !gst_rtsp_media_is_receive_only with
1888 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1889 for a sender pipeline that is complete. Only then a RTPInfo should
1891 gst_rtsp_media_is_receive_only gives different answears depending on
1893 If Describe is called wth URL+options for backchannel SDP will give only
1894 audio and only backchannel a=sendonly
1895 If Describe is called on URL+options that gives both audio and video
1896 direction from server to client, pipelines are created. Thus
1897 receive_only will return false, even though Setup only would setup
1899 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1900 streams are complete.
1902 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1904 * gst/rtsp-server/rtsp-client.c:
1905 * tests/check/gst/client.c:
1906 rtsp-client: RTP Info exists conditionally in PLAY
1907 If RTP Info is missing and it is not a receiver only, eg. audio
1908 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1909 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1910 Since 1.14 there is audio backchannel support. Thus RTP-info is
1911 conditional now. When audio backchannel only mode, there is no RTP-info.
1914 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1916 * examples/test-onvif-client.c:
1917 test-onvif-client: remove unused query
1919 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1921 * gst/rtsp-server/rtsp-client.c:
1922 rtsp-client: RTP Info must exist in PLAY response
1923 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1926 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1928 * examples/test-onvif-client.c:
1929 test-onvif-client: perform accurate seeks
1930 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1931 Also, modify how we compute the position: position queries in
1932 PAUSED mode fail to account for the newly-prerolled frame, leading
1933 to frame skips when performing seeks in that state. Instead,
1934 compute the current position from the last sample.
1936 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1938 * gst/rtsp-server/rtsp-client.c:
1939 * gst/rtsp-server/rtsp-media.c:
1940 * gst/rtsp-server/rtsp-media.h:
1941 * tests/check/gst/rtspserver.c:
1942 Use complete streams for scale and speed.
1943 Without this patch it's always stream0 that is used to get segment event
1944 that is used to set scale and speed. This even if client not doing SETUP
1945 for stream0. At least in suspend mode reset this not working since then
1946 it's just random if send_rtp_sink have got any segment event. There are
1947 no check if send_rtp_sink for stream0 got any data before media is
1948 prerolled after PLAY request.
1950 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1952 * examples/test-onvif-server.c:
1953 * examples/test-onvif-server.h:
1954 examples/onvif-server: fix werror build with clang
1955 ../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]
1956 self->incoming_segment->format, self->incoming_segment->flags,
1957 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1958 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1959 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1961 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1962 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1964 <scratch space>:77:1: note: expanded from here
1967 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1968 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1970 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1971 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1973 <scratch space>:9:1: note: expanded from here
1976 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
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>:12:1: note: expanded from here
1984 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1987 meson: Don't generate doc cache when no plugins are enabled
1988 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1990 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1992 * examples/test-onvif-client.c:
1993 test-onvif-client: stdin is not defined in MSVC
1995 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1997 * gst/rtsp-server/rtsp-media.c:
1998 rtsp-media: add missing Since tag
2000 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2002 * examples/test-onvif-client.c:
2003 test-onvif-client: STDIN_FILENO is not portable
2004 If not defined, define it to _fileno(stdin) on Windows, 0
2007 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2009 * examples/test-onvif-server.c:
2010 test-onvif-server: downgrade logging
2012 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2014 * examples/meson.build:
2015 * examples/test-onvif-client.c:
2016 * examples/test-onvif-server.c:
2017 examples: add ONVIF client / server example
2019 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2021 * gst/rtsp-server/rtsp-client.c:
2022 * gst/rtsp-server/rtsp-media.c:
2023 rtsp-client: define all seek accuracy flags from setup_play_mode
2024 We then pass those to adjust_play_mode, which needs to operate
2025 on the "final" seek flags, as previously the code in rtsp-media
2026 was assuming that accuracy seek flags (accurate / key_unit) should
2027 not be set if the flags passed to the seek method were already set.
2029 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
2031 * gst/rtsp-server/rtsp-media-factory-uri.c:
2032 * gst/rtsp-server/rtsp-media.c:
2033 rtsp-media: Try to get dynamic payloaders by name from their bin first
2034 First try "pay", then "pay_%s" (where %s == pad name). And only then
2035 fall back to the code that simply takes the first payloader that is
2037 The current code usually works (but is racy) because it will always take
2038 the payloader that was last added (due to g_list_prepend() when adding
2039 elements) in pad-added and that's usually the correct one. But if a new
2040 payloader is added between pad-added and us trying to get it, we would
2041 get the wrong payloader.
2043 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2045 * tests/check/gst/client.c:
2046 client test: expect any port in transport
2047 setup_multicast_client sets a 5000-5010 range for the client
2048 ports, it is incorrect to expect the transport to always use
2052 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2054 * tests/check/gst/onvif.c:
2055 onvif tests: use g_cond_wait() correctly
2056 g_cond_wait() has to be called in a loop until required conditions
2060 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
2062 * gst/rtsp-server/rtsp-stream.c:
2063 rtsp-stream: Not wait on receiver streams when pre-rolling
2064 Without this patch there are problem pre-rolling when using audio back
2066 Without this patch a probe will be created for all streams including
2067 the stream for audio backchannel. To pre-roll all this pads have to
2068 receive data. Since the stream for audio backchannel is a receiver this
2070 The solution is to never create any probes for streams that are for
2071 incomming data and instead set them as blocking already from beginning.
2073 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
2075 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2076 * gst/rtsp-server/rtsp-onvif-media.c:
2077 onvif-media: fix "void function returning a value" compiler warning
2079 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2081 * gst/rtsp-server/rtsp-media.c:
2082 rtsp-media: make sure streams are blocked when sending seek
2083 The recent ONVIF work exposed a race condition when dealing with
2084 multiple streams: one of the sinks may preroll before other streams
2085 have started flushing. This led to the pipeline posting async-done
2086 prematurely, when some streams were actually still in the middle
2087 of performing a flushing seek. The newly-added code looks up a
2088 sticky segment event on the first stream in order to respond to
2089 the PLAY request with accurate Scale and Speed headers. In the
2090 failure condition, the first stream was flushing, and thus had
2091 no sticky segment event, leading to the PLAY request failing,
2092 and in turn the test.
2094 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
2097 * gst/rtsp-server/rtsp-media-factory-uri.h:
2100 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2102 * gst/rtsp-server/rtsp-client.c:
2103 * gst/rtsp-server/rtsp-client.h:
2104 * gst/rtsp-server/rtsp-media.c:
2105 * gst/rtsp-server/rtsp-media.h:
2106 * gst/rtsp-server/rtsp-onvif-client.c:
2107 * gst/rtsp-server/rtsp-onvif-client.h:
2108 * gst/rtsp-server/rtsp-onvif-media-factory.c:
2109 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2110 * gst/rtsp-server/rtsp-onvif-media.c:
2111 * gst/rtsp-server/rtsp-onvif-server.h:
2112 * gst/rtsp-server/rtsp-stream.c:
2113 * gst/rtsp-server/rtsp-stream.h:
2114 * tests/check/gst/media.c:
2115 * tests/check/gst/onvif.c:
2116 * tests/check/meson.build:
2117 onvif: Implement and test the Streaming Specification
2118 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2120 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2122 * gst/rtsp-server/rtsp-client.c:
2123 * gst/rtsp-server/rtsp-client.h:
2124 rtsp-client: add gst_rtsp_client_get_stream_transport()
2125 This will be used in the onvif tests in order to validate the
2126 data transmitted over TCP: for streaming to continue after a
2127 data message has been provided to client->send_func, the client
2128 is responsible for marking the message as sent on the relevant
2131 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2133 * gst/rtsp-server/rtsp-client.c:
2134 client: Scale implies TRICK_MODE
2136 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2138 * gst/rtsp-server/rtsp-client.c:
2139 client: compare booleans, not pointers to them
2141 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
2143 * gst/rtsp-server/rtsp-media.c:
2144 * gst/rtsp-server/rtsp-stream.c:
2145 * tests/check/gst/media.c:
2146 Reverse playback support
2147 GStreamer plays segment from stop to start when doing reverse playback.
2148 RTSP implies that media should be played from start of Range header to
2149 its stop. Hence we swap start and stop times before passing them to
2151 Also make gst_rtsp_stream_query_stop always return value that can be
2152 used as stop time of Range header.
2154 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
2156 * gst/rtsp-server/rtsp-client.c:
2157 * gst/rtsp-server/rtsp-media.c:
2158 * gst/rtsp-server/rtsp-media.h:
2159 * tests/check/gst/client.c:
2160 rtsp-client: add support for Scale and Speed header
2161 Add support for the RTSP Scale and Speed headers by setting the rate in
2162 the seek to (scale*speed). We then check the resulting segment for rate
2163 and applied rate, and use them as values for the Speed and Scale headers
2165 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2167 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
2169 * gst/rtsp-server/rtsp-client.c:
2170 * gst/rtsp-server/rtsp-client.h:
2171 rtsp-client: allow sub classes to adjust the seek
2172 Adds a new virtual function, adjust_play_mode(), that allows
2173 sub classes to adjust the seek done on the media. The sub class can
2174 modify the values of the the seek flags and the rate.
2175 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2177 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
2179 * gst/rtsp-server/rtsp-media.c:
2180 * gst/rtsp-server/rtsp-media.h:
2181 * gst/rtsp-server/rtsp-stream.c:
2182 * gst/rtsp-server/rtsp-stream.h:
2183 * tests/check/gst/media.c:
2184 rtsp-media: allow specifying rate when seeking
2185 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
2186 caller to specify the rate for the seek. Also added functions in
2187 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
2188 https://bugzilla.gnome.org/show_bug.cgi?id=754575
2190 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
2194 meson: Bump minimal GLib version to 2.44
2195 This means we can use some newer features and get rid of some
2196 boilerplate code using the G_DECLARE_* macros.
2197 As discussed on IRC, 2.44 is old enough by now to start depending on it.
2199 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2201 * docs/libs/.gitignore:
2202 * docs/libs/Makefile.am:
2203 * docs/libs/gst-rtsp-server-docs.sgml:
2204 * docs/libs/gst-rtsp-server-sections.txt:
2205 * docs/libs/gst-rtsp-server.types:
2206 docs: remove obsolete gtk-doc related files
2208 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2210 * gst/rtsp-sink/gstrtspclientsink.c:
2211 doc: remove xml from comments
2213 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
2215 * docs/gst_plugins_cache.json:
2217 docs: Stop building the doc cache by default
2218 And update the cache
2219 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
2221 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
2223 * docs/gst_plugins_cache.json:
2224 docs: Update plugins documentation cache
2226 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
2229 * gst/rtsp-server/rtsp-context.c:
2230 * gst/rtsp-server/rtsp-session-pool.c:
2231 doc: Fix some docstrings
2233 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
2239 * docs/gst_plugins_cache.json:
2242 * docs/plugin-index.md:
2243 * docs/plugin-sitemap.txt:
2246 * docs/version.entities.in:
2247 * gst/rtsp-server/meson.build:
2248 * gst/rtsp-sink/meson.build:
2250 * meson_options.txt:
2251 docs: Port to hotdoc
2253 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2255 * gst/rtsp-server/rtsp-auth.c:
2256 * gst/rtsp-server/rtsp-client.h:
2257 rtsp-server: Fix various Since markers
2259 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
2261 * gst/rtsp-server/rtsp-media.c:
2262 * gst/rtsp-server/rtsp-sdp.c:
2263 * gst/rtsp-server/rtsp-session-media.c:
2264 * gst/rtsp-server/rtsp-stream.c:
2265 rtsp-server: Add various Since: 1.14 markers
2267 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
2269 * gst/rtsp-server/rtsp-media-factory.c:
2270 * gst/rtsp-server/rtsp-media.c:
2271 * gst/rtsp-server/rtsp-stream-transport.c:
2272 * gst/rtsp-server/rtsp-stream.c:
2273 rtsp-server: Add various missing Since: 1.16 markers
2275 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
2277 * gst/rtsp-sink/gstrtspclientsink.c:
2278 rtspclientsink: Set async-handling=false for the internal bins
2279 Without this we can easily run into a race condition with async state changes:
2280 - the pipeline is doing an async state change
2281 - we set the internal bins to PLAYING but that's ignored because an
2282 async state change is currently pending
2283 - the async state change finishes but does not change the state of the
2284 internal bins because of locked_state==TRUE
2285 - the internal bins stay in PAUSED forever
2287 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2289 * gst/rtsp-sink/gstrtspclientsink.c:
2290 rtspclientsink: Use write_messages() API to send buffer lists in one go
2291 And to write messages with multiple memories also via writev().
2293 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
2295 * gst/rtsp-server/rtsp-client.c:
2296 * gst/rtsp-server/rtsp-client.h:
2297 * gst/rtsp-server/rtsp-server-object.h:
2298 * gst/rtsp-server/rtsp-server.c:
2299 rtsp-client: Handle Content-Length limitation
2300 Add functionality to limit the Content-Length.
2301 API addition, Enhancement.
2302 Define an appropriate request size limit and reject requests
2303 exceeding the limit with response status 413 Request Entity Too Large
2306 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2313 === release 1.16.0 ===
2315 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2321 * gst-rtsp-server.doap:
2325 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
2327 * gst/rtsp-sink/gstrtspclientsink.c:
2328 rtspclientsink: Notify the stream transport about each written message
2329 Otherwise it will never try to send us the next one: it tries to keep
2330 exactly one message in-flight all the time.
2331 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
2332 in the client sink we always write data out synchronously.
2334 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
2336 * gst/rtsp-server/rtsp-stream.c:
2337 rtsp_server: Free thread pool before clean transport cache
2338 If not waiting for free thread pool before clean transport caches, there
2339 can be a crash if a thread is executing in transport list loop in
2340 function send_tcp_message.
2341 Also add a check if priv->send_pool in on_message_sent to avoid that a
2342 new thread is pushed during wait of free thread pool. This is possible
2343 since when waiting for free thread pool mutex have to be unlocked.
2345 === release 1.15.90 ===
2347 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
2353 * gst-rtsp-server.doap:
2357 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
2359 * gst/rtsp-server/rtsp-stream.c:
2360 rtsp-stream: Add support for GCM (RFC 7714)
2363 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
2365 * gst/rtsp-server/rtsp-session-pool.c:
2366 session pool: fix missing klass-> in klass->create_session
2368 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2371 g-i: pass --quiet to g-ir-scanner
2372 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
2373 that we get even if everything works just fine.
2374 We still get g-ir-scanner warnings and compiler warnings if
2375 we pass this option.
2377 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2380 g-i: silence 'nested extern' compiler warnings when building scanner binary
2381 We need a nested extern in our init section for the scanner binary
2382 so we can call gst_init to make sure GStreamer types are initialised
2383 (they are not all lazy init via get_type functions, but some are in
2384 exported variables). There doesn't seem to be any other mechanism to
2385 achieve this, so just remove that warning, it's not important at all.
2387 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
2390 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2392 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
2394 * gst/rtsp-server/rtsp-media.c:
2395 * tests/check/gst/media.c:
2396 rtsp-media: Handle set state when preparing.
2397 Handle the situation when a call to gst_rtsp_media_set_state is done
2398 when media status is preparing.
2399 Also add unit test for this scenario.
2400 The unit test simulate on a media level when two clients share a (live)
2402 Both clients have done SETUP and got responses. Now client 1 is doing
2403 play and client 2 is just closing the connection.
2404 Then without patch there are a problem when
2405 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
2406 And client2 is doing closing connection we can end up in a call
2407 to gst_rtsp_media_set_state when
2408 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
2409 shut down media is jumped over .
2410 With this patch and this scenario we wait until
2411 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
2412 execute after that and now we will execute the logic for
2415 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
2423 === release 1.15.2 ===
2425 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
2431 * gst-rtsp-server.doap:
2435 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
2437 * gst/rtsp-server/rtsp-media.c:
2438 * tests/check/gst/client.c:
2439 rtsp-media: Fix multicast use case with common media
2448 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
2450 * gst/rtsp-server/rtsp-client.c:
2451 * gst/rtsp-server/rtsp-stream.c:
2452 * gst/rtsp-server/rtsp-stream.h:
2453 rtsp-server: remove recursive behavior
2454 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2456 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2458 * gst/rtsp-server/rtsp-client.c:
2459 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
2460 And route all messages through the send_func if no send_messages_func
2462 We otherwise break backwards compatibility.
2464 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2466 * docs/libs/gst-rtsp-server-sections.txt:
2467 * gst/rtsp-server/rtsp-client.c:
2468 * gst/rtsp-server/rtsp-client.h:
2469 * gst/rtsp-server/rtsp-stream.c:
2470 rtsp-client: Add support for sending buffer lists directly
2471 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2473 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2475 * docs/libs/gst-rtsp-server-sections.txt:
2476 * gst/rtsp-server/rtsp-client.c:
2477 * gst/rtsp-server/rtsp-media.c:
2478 * gst/rtsp-server/rtsp-stream-transport.c:
2479 * gst/rtsp-server/rtsp-stream-transport.h:
2480 * gst/rtsp-server/rtsp-stream.c:
2481 * gst/rtsp-sink/gstrtspclientsink.c:
2482 rtsp-server: Add support for buffer lists
2483 This adds new functions for passing buffer lists through the different
2484 layers without breaking API/ABI, and enables the appsink to actually
2485 provide buffer lists.
2486 This should already reduce CPU usage and potentially context switches a
2487 bit by passing a whole buffer list from the appsink instead of
2488 individual buffers. As a next step it would be necessary to
2489 a) Add support for a vector of data for the GstRTSPMessage body
2490 b) Add support for sending multiple messages at once to the
2491 GstRTSPWatch and let it be handled internally
2492 c) Adding API to GOutputStream that works like writev()
2493 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2495 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
2497 * gst/rtsp-server/rtsp-client.c:
2498 client: Fix crash in close handler
2499 The close handler could trigger a crash because it invalidated the
2500 watch_context while still leaving a source attached to it which would be
2501 cleaned up at a later point.
2503 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
2505 * gst/rtsp-server/rtsp-stream.c:
2506 rtsp-stream: Use cached address when allocating sockets
2507 If an address/port was previously decided upon (ex: multicast in the
2508 SDP), then use that instead of re-creating another one
2509 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2511 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
2513 * gst/rtsp-server/rtsp-media.c:
2514 rtsp-media: Fix race codition in finish_unprepare
2515 The previous fix for race condition around finish_unprepare where the
2516 function could be called twice assumed that the status wouldn't change
2517 during execution of the function. This assumption is incorrect as the
2518 state may change, for example if an error message arrives from the
2520 Instead a flag keeping track on whether the finish_unprepare function
2521 is currently executing is introduced and checked.
2522 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2524 === release 1.15.1 ===
2526 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2532 * gst-rtsp-server.doap:
2536 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
2538 * gst/rtsp-server/rtsp-stream.c:
2539 Add source elements to the pipeline before activation
2540 In plug_src we changed the element state before adding it to
2541 the owner container. This prevented the pipeline from intercepting
2542 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
2543 to assign a custom task pool.
2544 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2546 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
2549 Automatic update of common submodule
2550 From ed78bee to 59cb678
2552 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
2554 * examples/test-appsrc.c:
2555 examples: test-appsrc: fix coding style error
2557 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
2559 * examples/test-appsrc.c:
2560 examples: test-appsrc: fix buffer leak
2562 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
2564 * gst/rtsp-server/rtsp-media.c:
2565 rtsp-media: Update priv->blocked when linked streams are unblocked.
2566 Media is considered to be blocked when all streams that belong to
2567 that media are blocked.
2568 This patch solves the problem of inconsistent updates of
2569 priv->blocked that are not synchronized with the media state.
2571 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
2573 * gst/rtsp-server/rtsp-media.c:
2574 rtsp-media: Don't block streams before seeking
2575 Before the seek operation is performed on media, it's required that
2576 its pipeline is prepared <=> the pipeline is in the PAUSED state.
2577 At this stage, all transport parts (transport sinks) have been successfully
2578 added to the pipeline and there is no need for blocking the streams.
2580 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
2582 * tests/check/gst/rtspserver.c:
2583 tests: rtspserver: Add shared media test case for TCP
2585 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
2587 * gst/rtsp-server/rtsp-stream.c:
2588 rtsp-stream: Use seqnum-offset for rtpinfo
2589 The sequence number in the rtpinfo is supposed to be the first RTP
2590 sequence number. The "seqnum" property on a payloader is supposed to be
2591 the number from the last processed RTP packet. The sequence number for
2592 payloaders that inherit gstrtpbasepayload will not be correct in case of
2593 buffer lists. In order to fix the seqnum property on the payloaders
2594 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
2595 "seqnum-offset" from the "stats" property contains the value of the
2596 very first RTP packet in a stream. The server will, however, try to look
2597 at the last simple in the sink element and only use properties on the
2598 payloader in case there no sink elements yet, and by looking at the last
2599 sample of the sink gives the server full control of which RTP packet it
2600 looks at. If the payloader does not have the "stats" property, "seqnum"
2601 is still used since "seqnum-offset" is only present in as part of
2602 "stats" and this is still an issue not solved with this patch.
2603 Needed for gst-plugins-base!17
2605 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2607 * gst/rtsp-server/rtsp-stream.c:
2608 rtsp-stream: Plug memory leak
2609 Attaching a GSource to a context will increase the refcount. The idle
2610 source will never be free'd since the initial reference is never
2613 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2616 Add Gitlab CI configuration
2617 This commit adds a .gitlab-ci.yml file, which uses a feature
2618 to fetch the config from a centralized repository. The intent is
2619 to have all the gstreamer modules use the same configuration.
2620 The configuration is currently hosted at the gst-ci repository
2621 under the gitlab/ci_template.yml path.
2622 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2624 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2627 * gst-rtsp-server.doap:
2628 Update git locations to gitlab
2630 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2632 * gst/rtsp-server/meson.build:
2633 meson: add new onvif types
2635 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2637 * gst/rtsp-server/meson.build:
2638 Add ONVIF subclass headers to the installed headers in meson.build too
2640 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2642 * gst/rtsp-server/rtsp-server-object.h:
2643 * gst/rtsp-server/rtsp-server.h:
2644 rtsp-server: Declare GstRTSPServer struct before anything else
2645 It's needed by all kinds of other headers, including the ones that are
2646 required for defining the GstRTSPServer struct itself and its API.
2648 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2650 * gst/rtsp-server/rtsp-onvif-client.h:
2651 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2652 * gst/rtsp-server/rtsp-onvif-media.h:
2653 * gst/rtsp-server/rtsp-onvif-server.h:
2654 Mark all ONVIF-specific subclasses as Since 1.14
2656 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2658 * gst/rtsp-server/Makefile.am:
2659 * gst/rtsp-server/meson.build:
2660 * gst/rtsp-server/rtsp-context.h:
2661 * gst/rtsp-server/rtsp-onvif-server.c:
2662 * gst/rtsp-server/rtsp-onvif-server.h:
2663 * gst/rtsp-server/rtsp-server-object.h:
2664 * gst/rtsp-server/rtsp-server-prelude.h:
2665 * gst/rtsp-server/rtsp-server.c:
2666 * gst/rtsp-server/rtsp-server.h:
2667 * gst/rtsp-server/rtsp-session.h:
2668 Include ONVIF types from single-include rtsp-server.h
2669 ... by actually making it a single-include header and moving everything
2670 related to the GstRTSPServer type to rtsp-server-object.h instead.
2671 Otherwise there are too many circular includes.
2672 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2674 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2676 * gst/rtsp-server/rtsp-client.c:
2677 * gst/rtsp-server/rtsp-latency-bin.c:
2678 * gst/rtsp-server/rtsp-stream.c:
2679 * gst/rtsp-server/rtsp-stream.h:
2680 rtsp-stream: use idle source in on_message_sent
2681 When the underlying layers are running on_message_sent, this sometimes
2682 causes the underlying layer to send more data, which will cause the
2683 underlying layer to run callback on_message_sent again. This can go on
2685 To break this chain, we introduce an idle source that takes care of
2686 sending data if there are more to send when running callback
2687 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2689 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2691 * gst/rtsp-server/rtsp-client.c:
2692 rtsp-client: Remove timeout GSource on cleanup
2693 Avoids ending up with races where a timeout would still be around
2694 *after* a client was gone. This could happen rather easily in
2695 RTSP-over-HTTP mode on a local connection, where each RTSP message
2696 would be sent as a different HTTP connection with the same tunnelid.
2697 If not properly removed, that timeout would then try to free again
2698 a client (and its contents).
2700 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2702 * gst/rtsp-server/Makefile.am:
2703 autotools: fix distcheck
2705 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2707 * gst/rtsp-server/Makefile.am:
2708 * gst/rtsp-server/meson.build:
2709 * gst/rtsp-server/rtsp-latency-bin.c:
2710 * gst/rtsp-server/rtsp-latency-bin.h:
2711 * gst/rtsp-server/rtsp-onvif-media.c:
2712 onvif: encapsulate onvif part into a bin
2713 ...and thus do not let onvif affect pipelines latency
2714 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2716 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2718 * tests/check/gst/client.c:
2719 tests: client: Avoid bind() failures in tests
2720 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2722 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2724 * gst/rtsp-server/rtsp-media-factory.c:
2725 * gst/rtsp-server/rtsp-media-factory.h:
2726 * gst/rtsp-server/rtsp-media.c:
2727 * gst/rtsp-server/rtsp-media.h:
2728 * gst/rtsp-server/rtsp-stream.c:
2729 * gst/rtsp-server/rtsp-stream.h:
2730 * tests/check/gst/client.c:
2731 * tests/check/gst/mediafactory.c:
2732 New property for socket binding to mcast addresses
2733 By default the multicast sockets are bound to INADDR_ANY,
2734 as it's not allowed to bind sockets to multicast addresses
2735 in Windows. This default behaviour can be changed by setting
2736 bind-mcast-address property on the media-factory object.
2737 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2739 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2742 * gst/rtsp-server/Makefile.am:
2743 * gst/rtsp-server/meson.build:
2744 * gst/rtsp-server/rtsp-address-pool.c:
2745 * gst/rtsp-server/rtsp-auth.c:
2746 * gst/rtsp-server/rtsp-client.c:
2747 * gst/rtsp-server/rtsp-context.c:
2748 * gst/rtsp-server/rtsp-media-factory-uri.c:
2749 * gst/rtsp-server/rtsp-media-factory.c:
2750 * gst/rtsp-server/rtsp-media.c:
2751 * gst/rtsp-server/rtsp-mount-points.c:
2752 * gst/rtsp-server/rtsp-params.c:
2753 * gst/rtsp-server/rtsp-permissions.c:
2754 * gst/rtsp-server/rtsp-sdp.c:
2755 * gst/rtsp-server/rtsp-server-prelude.h:
2756 * gst/rtsp-server/rtsp-server.c:
2757 * gst/rtsp-server/rtsp-session-media.c:
2758 * gst/rtsp-server/rtsp-session-pool.c:
2759 * gst/rtsp-server/rtsp-session.c:
2760 * gst/rtsp-server/rtsp-stream-transport.c:
2761 * gst/rtsp-server/rtsp-stream.c:
2762 * gst/rtsp-server/rtsp-thread-pool.c:
2763 * gst/rtsp-server/rtsp-token.c:
2765 libs: fix API export/import and 'inconsistent linkage' on MSVC
2766 Export rtsp-server library API in headers when we're building the
2767 library itself, otherwise import the API from the headers.
2768 This fixes linker warnings on Windows when building with MSVC.
2769 Fix up some missing config.h includes when building the lib which
2770 is needed to get the export api define from config.h
2771 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2773 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2775 * gst/rtsp-server/rtsp-media-factory.c:
2776 rtsp-media-factory: Add missing break statements
2777 This resulted in warnings/assertions whenever one accessed the
2778 max-mcast-ttl property.
2782 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2785 * meson_options.txt:
2786 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2788 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2791 * meson_options.txt:
2792 * tests/check/meson.build:
2793 meson: add option to disable build of rtspclientsink plugin
2795 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2797 * meson_options.txt:
2798 meson: re-arrange options
2800 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2803 * meson_options.txt:
2804 * tests/check/meson.build:
2805 * tests/meson.build:
2806 meson: Use feature option for tests option
2807 This was somehow missed the last time around.
2809 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2811 * gst/rtsp-server/meson.build:
2813 meson: Maintain macOS ABI through dylib versioning
2814 Requires Meson 0.48, but the feature will be ignored on older versions
2815 so it's safe to add it without bumping the requirement.
2817 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2819 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2821 * gst/rtsp-sink/meson.build:
2823 meson: add pkg-config file for the rtspclientsink plugin
2825 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2827 * gst/rtsp-server/rtsp-client.c:
2828 * tests/check/gst/client.c:
2829 rtsp-client: Avoid reuse of channel numbers for interleaved
2830 If a (strange) client would reuse interleaved channel numbers in
2831 multiple SETUP requests, we should not accept them. The channel
2832 numbers are used for looking up stream transports in the
2833 priv->transports hash table, and transports disappear from the table
2834 if channel numbers are reused.
2835 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2836 server to change the channel numbers suggested by the client.
2837 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2839 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2841 * tests/check/gst/client.c:
2842 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2843 Allow regex for matching transport header against expected pattern.
2844 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2846 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2848 * tests/check/meson.build:
2849 meson: There is no gstreamer-plugins-good-1.0.pc
2850 There is no installed version of that, only an uninstalled version.
2852 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2854 * gst/rtsp-server/rtsp-client.c:
2855 * tests/check/gst/stream.c:
2856 Fix indentation again
2858 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2860 * gst/rtsp-server/rtsp-client.c:
2861 * gst/rtsp-server/rtsp-stream.c:
2862 * gst/rtsp-server/rtsp-stream.h:
2863 * tests/check/gst/client.c:
2864 * tests/check/gst/stream.c:
2865 stream: Added a list of multicast client addresses
2866 When media is shared, the same media stream can be sent
2867 to multiple multicast groups. Currently, there is no API
2868 to retrieve multicast addresses from the stream.
2869 When calling gst_rtsp_stream_get_multicast_address() function,
2870 only the first multicast address is returned.
2871 With this patch, each multicast destination requested in SETUP
2872 will be stored in an internal list (call to
2873 gst_rtsp_stream_add_multicast_client_address()).
2874 The list of multicast groups requested by the clients can be
2875 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2876 There still exist some problems with the current implementation
2877 in the multicast case:
2878 1) The receiving part is currently only configured with
2879 regard to the first multicast client (see
2880 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2881 2) Secondly, of security reasons, some constraints should be
2882 put on the requested multicast destinations (see
2883 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2884 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2885 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2887 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2889 * gst/rtsp-server/rtsp-client.c:
2890 * gst/rtsp-server/rtsp-stream.c:
2891 * gst/rtsp-server/rtsp-stream.h:
2892 * tests/check/gst/client.c:
2893 stream: Choose the maximum ttl value provided by multicast clients
2894 The maximum ttl value provided so far by the multicast clients
2895 will be chosen and reported in the response to the current
2897 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2898 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2900 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2902 * gst/rtsp-server/rtsp-stream.c:
2903 * tests/check/gst/client.c:
2904 rtsp-stream: Don't require address pool in the transport specific case
2905 If "transport.client-settings" parameter is set to true, the client is
2906 allowed to specify destination, ports and ttl.
2907 There is no need for pre-configured address pool.
2908 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2909 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2911 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2913 * gst/rtsp-server/rtsp-client.c:
2914 * tests/check/gst/client.c:
2915 client: Don't reserve multicast address in the client setting case
2916 When two multicast clients request specific transport
2917 configurations, and "transport.client-settings" parameter is
2918 set to true, it's wrong to actually require that these two
2919 clients request the same multicast group.
2920 Removed test_client_multicast_invalid_transport_specific test
2921 cases as they wrongly require that the requested destination
2922 address is supposed to be present in the address pool, also in
2923 the case when "transport.client-settings" parameter is set to true.
2924 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2925 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2927 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2929 * gst/rtsp-server/rtsp-media-factory.c:
2930 * gst/rtsp-server/rtsp-media-factory.h:
2931 * gst/rtsp-server/rtsp-media.c:
2932 * gst/rtsp-server/rtsp-media.h:
2933 * gst/rtsp-server/rtsp-stream.c:
2934 * gst/rtsp-server/rtsp-stream.h:
2935 * tests/check/gst/mediafactory.c:
2936 Add new API for setting/getting maximum multicast ttl value
2937 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2938 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2940 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2942 * gst/rtsp-server/rtsp-stream.c:
2943 rtsp-stream: avoid duplicating the first multicast client
2944 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2945 clients were dynamically added and removed to the multicast
2946 udp sinks, as such we should no longer add a first client in
2947 set_multicast_socket_for_udpsink
2948 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2950 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2952 * gst/rtsp-server/rtsp-stream.c:
2953 Revert "rtsp-stream: avoid duplicating the first multicast client"
2954 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2955 Commits where accidentially squashed together
2957 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2959 * gst/rtsp-server/rtsp-client.c:
2960 * gst/rtsp-server/rtsp-media-factory.c:
2961 * gst/rtsp-server/rtsp-media-factory.h:
2962 * gst/rtsp-server/rtsp-media.c:
2963 * gst/rtsp-server/rtsp-media.h:
2964 * gst/rtsp-server/rtsp-stream.c:
2965 * gst/rtsp-server/rtsp-stream.h:
2966 * tests/check/gst/client.c:
2967 * tests/check/gst/mediafactory.c:
2968 Revert "Add new API for setting/getting maximum multicast ttl value"
2969 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2970 Commits where accidentially squashed together
2972 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2974 * gst/rtsp-server/rtsp-stream.c:
2975 * tests/check/gst/client.c:
2976 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2977 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2978 Commits where accidentially squashed together
2980 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2982 * gst/rtsp-server/rtsp-client.c:
2983 * gst/rtsp-server/rtsp-stream.c:
2984 * gst/rtsp-server/rtsp-stream.h:
2985 * tests/check/gst/client.c:
2986 * tests/check/gst/stream.c:
2987 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2988 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2989 Commits where accidentially squashed together
2991 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2993 * examples/test-auth-digest.c:
2994 examples: Fix indentation
2996 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.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 stream: Choose the maximum ttl value provided by multicast clients
3004 The maximum ttl value provided so far by the multicast clients
3005 will be chosen and reported in the response to the current
3007 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3009 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3011 * gst/rtsp-server/rtsp-stream.c:
3012 * tests/check/gst/client.c:
3013 rtsp-stream: Don't require address pool in the transport specific case
3014 If "transport.client-settings" parameter is set to true, the client is
3015 allowed to specify destination, ports and ttl.
3016 There is no need for pre-configured address pool.
3017 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3019 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
3021 * gst/rtsp-server/rtsp-client.c:
3022 * gst/rtsp-server/rtsp-media-factory.c:
3023 * gst/rtsp-server/rtsp-media-factory.h:
3024 * gst/rtsp-server/rtsp-media.c:
3025 * gst/rtsp-server/rtsp-media.h:
3026 * gst/rtsp-server/rtsp-stream.c:
3027 * gst/rtsp-server/rtsp-stream.h:
3028 * tests/check/gst/client.c:
3029 * tests/check/gst/mediafactory.c:
3030 Add new API for setting/getting maximum multicast ttl value
3031 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3033 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3035 * gst/rtsp-server/rtsp-stream.c:
3036 rtsp-stream: avoid duplicating the first multicast client
3037 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
3038 clients were dynamically added and removed to the multicast
3039 udp sinks, as such we should no longer add a first client in
3040 set_multicast_socket_for_udpsink
3041 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3043 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
3045 * gst/rtsp-server/Makefile.am:
3046 rtsp-server: Add gstreamer-base gir dir in autotools
3048 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3050 * gst/rtsp-server/rtsp-client.c:
3051 * gst/rtsp-server/rtsp-stream.c:
3052 rtsp-client: always allocate both IPV4 and IPV6 sockets
3053 multiudpsink does not support setting the socket* properties
3054 after it has started, which meant that rtsp-server could no
3055 longer serve on both IPV4 and IPV6 sockets since the patches
3056 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
3058 When first connecting an IPV6 client then an IPV4 client,
3059 multiudpsink fell back to using the IPV6 socket.
3060 When first connecting an IPV4 client, then an IPV6 client,
3061 multiudpsink errored out, released the IPV4 socket, then
3062 crashed when trying to send a message on NULL nevertheless,
3063 that is however a separate issue.
3064 This could probably be fixed by handling the setting of
3065 sockets in multiudpsink after it has started, that will
3066 however be a much more significant effort.
3067 For now, this commit simply partially reverts the behaviour
3068 of rtsp-stream: it will continue to only create the udpsinks
3069 when needed, as was the case since the patches were merged,
3070 it will however when creating them, always allocate both
3071 sockets and set them on the sink before it starts, as was
3072 the case prior to the patches.
3073 Transport configuration will only error out if the allocation
3074 of UDP sockets fails for the actual client's family, this
3075 also downgrades the GST_ERRORs in alloc_ports_one_family
3076 to GST_WARNINGs, as failing to allocate is no longer
3078 https://bugzilla.gnome.org/show_bug.cgi?id=796875
3080 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3083 * meson_options.txt:
3084 meson: Convert common options to feature options
3085 These are necessary for gst-build to set options correctly. The
3086 remaining automagic option is cgroup support in examples.
3087 https://bugzilla.gnome.org/show_bug.cgi?id=795107
3089 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
3091 * gst/rtsp-server/rtsp-stream.c:
3092 rtsp-stream: Slightly simplify locking
3094 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
3096 * gst/rtsp-server/rtsp-client.c:
3097 * gst/rtsp-server/rtsp-stream-transport.c:
3098 * gst/rtsp-server/rtsp-stream-transport.h:
3099 * gst/rtsp-server/rtsp-stream.c:
3100 Limit queued TCP data messages to one per stream
3101 Before, the watch backlog size in GstRTSPClient was changed
3102 dynamically between unlimited and a fixed size, trying to avoid both
3103 unlimited memory usage and deadlocks while waiting for place in the
3104 queue. (Some of the deadlocks were described in a long comment in
3106 In the previous commit, we changed to a fixed backlog size of 100.
3107 This is possible, because we now handle RTP/RTCP data messages differently
3108 from RTSP request/response messages.
3109 The data messages are messages tunneled over TCP. We allow at most one
3110 queued data message per stream in GstRTSPClient at a time, and
3111 successfully sent data messages are acked by sending a "message-sent"
3112 callback from the GstStreamTransport. Until that ack comes, the
3113 GstRTSPStream does not call pull_sample() on its appsink, and
3114 therefore the streaming thread in the pipeline will not be blocked
3115 inside GstRTSPClient, waiting for a place in the queue.
3116 pull_sample() is called when we have both an ack and a "new-sample"
3117 signal from the appsink. Then, we know there is a buffer to write.
3118 RTSP request/response messages are not acked in the same way as data
3119 messages. The rest of the 100 places in the queue are used for
3120 them. If the queue becomes full of request/response messages, we
3121 return an error and close the connection to the client.
3122 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
3124 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
3126 * gst/rtsp-server/rtsp-client.c:
3127 rtsp-client: Use fixed backlog size
3128 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
3129 Preparation for the next commit, which changes to a different way of
3130 avoiding both deadlocks and unlimited memory usage with the watch
3133 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3135 * gst/rtsp-server/rtsp-media.c:
3136 rtsp-media: unref clock (if set) when finalizing
3137 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3139 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
3141 * docs/libs/gst-rtsp-server-sections.txt:
3142 rtsp-media: add gst_rtsp_media_*_set_clock to docs
3143 https://bugzilla.gnome.org/show_bug.cgi?id=796814
3145 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
3147 * gst/rtsp-server/rtsp-media-factory.c:
3148 media-factory: unref old clock when setting new clock
3149 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3151 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
3153 * gst/rtsp-server/rtsp-media-factory.c:
3154 media-factory: unref clock in finalize
3155 https://bugzilla.gnome.org/show_bug.cgi?id=796724
3157 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
3159 * gst/rtsp-server/rtsp-onvif-media.c:
3160 rtsp-onvif-media: fix g-ir-scanner warnings
3162 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3165 .gitignore: add another example binary
3167 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
3169 * examples/meson.build:
3170 meson: add new test-appsrc2 example to meson build
3172 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
3174 * examples/Makefile.am:
3175 examples: fix build of new test-appsrc2 example
3176 Need to link against libgstapp-1.0.
3178 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
3180 * examples/.gitignore:
3181 * examples/Makefile.am:
3182 * examples/test-appsrc2.c:
3183 examples: Add test-appsrc2
3184 Add an example of feeding both audio and video into an RTSP
3185 pipeline via appsrc.
3187 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
3189 * gst/rtsp-server/rtsp-client.c:
3190 client: Strip transport parts as whitespaces could be around commas
3191 https://bugzilla.gnome.org/show_bug.cgi?id=758428
3193 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
3195 * gst/rtsp-server/rtsp-stream.c:
3196 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
3197 Fix race when setting up source elements.
3198 Since we set the source element(s) to PLAYING state before hooking
3199 them up to the downstream funnel, it's possible for the source element
3200 to receive packets before we actually get to linking it to the funnel,
3201 in which case buffers would be pushed out on an unlinked pad, causing
3202 it to error out and stop receiving more data.
3203 We fix this by blocking the source's srcpad until we have linked it.
3204 https://bugzilla.gnome.org/show_bug.cgi?id=796160
3206 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
3208 * gst/rtsp-server/rtsp-stream.c:
3209 rtsp-stream: Fix mismatch between allowed and configured protocols
3210 https://bugzilla.gnome.org/show_bug.cgi?id=796679
3212 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
3214 * gst/rtsp-server/rtsp-stream.c:
3215 rtsp-stream: Emit a signal when the SRTP decoder is created
3216 https://bugzilla.gnome.org/show_bug.cgi?id=778080
3218 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
3220 * gst/rtsp-server/rtsp-stream.c:
3221 rtsp-stream: Don't require presence of sinks in _get_*_socket()
3222 Transport specific sink elements are added to the pipeline
3223 in PLAY request and sockets are already created in SETUP so
3224 it's actually wrong to require the presence of sinks in
3225 _get_*_socket() functions.
3226 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3228 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
3230 * gst/rtsp-server/rtsp-stream.c:
3231 rtsp-stream: Update transport for multicast clients as well
3232 If a multicast client requests different transport settings
3233 than the existing one make sure that this new transport
3234 configuruation is propagated to the multicast udp sink.
3235 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3237 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
3239 * gst/rtsp-server/rtsp-stream.c:
3240 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
3241 And not on unicast udp sinks
3242 https://bugzilla.gnome.org/show_bug.cgi?id=793441
3244 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
3246 * gst/rtsp-server/rtsp-address-pool.c:
3247 * gst/rtsp-server/rtsp-auth.c:
3248 * gst/rtsp-server/rtsp-client.c:
3249 * gst/rtsp-server/rtsp-media-factory-uri.c:
3250 * gst/rtsp-server/rtsp-media-factory.c:
3251 * gst/rtsp-server/rtsp-media.c:
3252 * gst/rtsp-server/rtsp-mount-points.c:
3253 * gst/rtsp-server/rtsp-server.c:
3254 * gst/rtsp-server/rtsp-session-media.c:
3255 * gst/rtsp-server/rtsp-session-pool.c:
3256 * gst/rtsp-server/rtsp-session.c:
3257 * gst/rtsp-server/rtsp-stream-transport.c:
3258 * gst/rtsp-server/rtsp-stream.c:
3259 * gst/rtsp-server/rtsp-thread-pool.c:
3260 Update for g_type_class_add_private() deprecation in recent GLib
3262 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
3264 * gst/rtsp-server/rtsp-auth.c:
3265 * gst/rtsp-server/rtsp-media.c:
3266 * gst/rtsp-server/rtsp-sdp.c:
3267 * gst/rtsp-server/rtsp-stream.c:
3270 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
3272 * examples/Makefile.am:
3273 * examples/test-video-disconnect.c:
3274 examples: Add test-video-disconnect example
3275 Simple example which cuts off all clients 10 seconds
3276 after the first one connects.
3278 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3280 * docs/libs/gst-rtsp-server-sections.txt:
3281 * examples/test-auth-digest.c:
3282 * gst/rtsp-server/rtsp-auth.c:
3283 * gst/rtsp-server/rtsp-auth.h:
3284 rtsp-auth: Add support for parsing .htdigest files
3285 Passwords are usually not stored in clear text, but instead
3286 stored already hashed in a .htdigest file.
3287 Add support for parsing such files, add API to allow setting
3288 a custom realm in RTSPAuth, and update the digest example.
3289 https://bugzilla.gnome.org/show_bug.cgi?id=796637
3291 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
3293 * gst/rtsp-sink/gstrtspclientsink.c:
3294 * gst/rtsp-sink/gstrtspclientsink.h:
3295 rtspclientsink: fix waiting for multiple streams
3296 We were previously only ever waiting for a single stream to notify it's
3297 blocked status through GstRTSPStreamBlocking. Actually count streams to
3299 Fixes rtspclientsink sending SDP's without out some of the input
3301 https://bugzilla.gnome.org/show_bug.cgi?id=796624
3303 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3305 * docs/libs/gst-rtsp-server-sections.txt:
3306 docs: add missing auth methods
3308 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3310 * gst/rtsp-server/rtsp-stream.c:
3311 rtsp-stream: only create funnel if it didn't exist already.
3312 This precented using multiple protocols for the same stream.
3313 https://bugzilla.gnome.org/show_bug.cgi?id=796634
3315 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3317 * examples/meson.build:
3318 meson: build auth-digest example
3320 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
3322 * gst/rtsp-server/rtsp-client.c:
3323 * gst/rtsp-server/rtsp-media.c:
3324 * gst/rtsp-server/rtsp-sdp.c:
3325 * gst/rtsp-server/rtsp-session-media.c:
3326 * gst/rtsp-server/rtsp-stream-transport.c:
3327 Get payloader stats only for the sending streams
3328 Get/set payloader properties only for streams that actually
3329 contain a payloader element.
3330 https://bugzilla.gnome.org/show_bug.cgi?id=796523
3332 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
3334 * gst/rtsp-server/Makefile.am:
3335 Makefile: Don't hardcode libtool for g-i build
3336 Similar to the other commits in core/base/bad
3338 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
3340 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3341 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
3342 https://bugzilla.gnome.org/show_bug.cgi?id=796229
3344 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
3346 * gst/rtsp-sink/gstrtspclientsink.c:
3347 rtspclientsink: Don't deadlock in preroll on early close
3348 If the connection is closed very early, the flushing
3349 marker might not get set and rtspclientsink can get
3350 deadlocked waiting for preroll forever.
3351 https://bugzilla.gnome.org/show_bug.cgi?id=786961
3353 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3356 * meson_options.txt:
3357 meson: Update option names to omit disable_ and with- prefixes
3358 Also yield common options to the outer project (gst-build in our case)
3359 so that they don't have to be set manually.
3361 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3364 meson: use -Wl,-Bsymbolic-functions where supported
3365 Just like the autotools build.
3367 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3370 * tests/check/Makefile.am:
3371 configure: check for -good and -bad plugins only in uninstalled setup
3372 Avoids confusing configure messages looking or a -good .pc file
3374 Also use plugindir variables that common macros set while at it.
3375 https://bugzilla.gnome.org/show_bug.cgi?id=795466
3377 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
3379 * gst/rtsp-server/rtsp-client.c:
3380 rtsp-client: Fix session timeout
3381 When streaming data over TCP then is not the keep-alive
3382 functionality working.
3383 The reason is that the function do_send_data have changed
3384 to boolean but the code is still checking the received result
3385 from send_func with GST_RTSP_OK.
3386 The result is that a successful send_func will always lead to
3387 that do_send_data is returning false and the keep-alive will
3389 https://bugzilla.gnome.org/show_bug.cgi?id=795321
3391 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3393 * docs/libs/gst-rtsp-server-sections.txt:
3394 * gst/rtsp-server/rtsp-media.c:
3395 * gst/rtsp-server/rtsp-sdp.c:
3396 * gst/rtsp-server/rtsp-stream.c:
3397 * gst/rtsp-server/rtsp-stream.h:
3398 * gst/rtsp-sink/gstrtspclientsink.c:
3399 * gst/rtsp-sink/gstrtspclientsink.h:
3400 Implement support for ULP Forward Error Correction
3401 In this initial commit, interface is only exposed for RECORD,
3402 further work will be needed in rtspsrc to support this for
3404 https://bugzilla.gnome.org/show_bug.cgi?id=794911
3406 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
3408 * gst/rtsp-server/rtsp-onvif-media.c:
3409 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
3410 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
3411 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
3412 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
3413 the opposite, just like the ONVIF standard.
3414 Let's follow those RFCs as we're doing RTSP here, and add a property at
3415 a later time if needed to switch to the SDP RFC behaviour.
3416 https://bugzilla.gnome.org/show_bug.cgi?id=793964
3418 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
3421 Automatic update of common submodule
3422 From 3fa2c9e to ed78bee
3424 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
3426 * gst/rtsp-server/rtsp-client.c:
3427 * gst/rtsp-server/rtsp-media-factory.c:
3428 * gst/rtsp-server/rtsp-media.c:
3429 * gst/rtsp-server/rtsp-stream.c:
3430 * tests/check/gst/rtspclientsink.c:
3431 gst: Run everything through gst-indent again
3433 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
3435 * gst/rtsp-server/rtsp-media.c:
3436 * tests/check/gst/media.c:
3437 rtsp-media: query the position on active streams if media is complete
3438 If the media is complete, i.e. one or more streams have been configured
3439 with sinks, then we want to query the position on those streams only.
3440 A query on an incomplete stream may return a position that originates from
3442 https://bugzilla.gnome.org/show_bug.cgi?id=794964
3444 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3446 * gst/rtsp-sink/gstrtspclientsink.c:
3447 rtspclientsink: make sure not to use freed string
3448 Set transport string to NULL after freeing it, so that
3449 at worst we get a NULL pointer if constructing a new
3450 transport string fails (which shouldn't really fail here).
3451 Also check return value of that, just in case.
3454 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3456 * gst/rtsp-server/rtsp-client.c:
3457 rtsp-client: do not free string passed to take_header
3459 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3461 * gst/rtsp-server/rtsp-stream.c:
3462 rtsp-stream: do not take lock in request_aux_receiver
3463 Added it right before pushing the previous commit, it is
3464 incorrect and deadlocks because this function gets called
3465 from the join_bin thread, which already holds the lock,
3466 that's the reason why request_aux_sender didn't take the
3469 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3471 * docs/libs/gst-rtsp-server-sections.txt:
3472 * gst/rtsp-server/rtsp-media-factory.c:
3473 * gst/rtsp-server/rtsp-media-factory.h:
3474 * gst/rtsp-server/rtsp-media.c:
3475 * gst/rtsp-server/rtsp-media.h:
3476 * gst/rtsp-server/rtsp-stream.c:
3477 * gst/rtsp-server/rtsp-stream.h:
3478 rtsp-server: add API to enable retransmission requests
3479 "do-retransmission" was previously set when rtx-time != 0,
3480 which made no sense as do-retransmission is used to enable
3481 the sending of retransmission requests, where as rtx-time
3482 is used by the peer to enable storing of buffers in order
3483 to respond to retransmission requests.
3484 rtsp-media now also provides a callback for the
3485 request-aux-receiver signal.
3486 https://bugzilla.gnome.org/show_bug.cgi?id=794822
3488 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3490 * gst/rtsp-sink/gstrtspclientsink.c:
3491 rtspclientsink: add rtx ssrc to mikey's crypto sessions
3492 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3494 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3496 * gst/rtsp-sink/gstrtspclientsink.c:
3497 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
3498 This in order to be able to decrypt the RTCP backchannel
3499 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3501 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3503 * gst/rtsp-server/rtsp-client.c:
3504 rtsp-client: Send KeyMgmt header in ANNOUNCE response
3505 When sending back an encrypted RTCP back channel, it is useful
3506 for the client to know the encryption key.
3507 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3509 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3511 * gst/rtsp-server/rtsp-client.c:
3512 * gst/rtsp-server/rtsp-stream.c:
3513 * gst/rtsp-server/rtsp-stream.h:
3514 rtsp-stream: extract handle_keymgmt from rtsp-client
3515 rtspclientsink will also need to parse KeyMgmt headers
3516 sent by the server to decrypt the RTCP backchannel stream
3517 https://bugzilla.gnome.org/show_bug.cgi?id=794813
3519 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3521 * gst/rtsp-sink/gstrtspclientsink.c:
3522 * tests/check/gst/rtspclientsink.c:
3523 rtspclientsink: Fix client ports for the RTCP backchannel
3524 This was broken since the work for delayed transport creation
3525 was merged: the creation of the transports string depends on
3526 calling stream_get_server_port, which only starts returning
3527 something meaningful after a call to stream_allocate_udp_sockets
3528 has been made, this function expects a transport that we parse
3529 from the transport string ...
3530 Significant refactoring is in order, but does not look entirely
3531 trivial, for now we put a band aid on and create a second transport
3532 string after the stream has been completed, to pass it in
3533 the request headers instead of the previous, incomplete one.
3534 https://bugzilla.gnome.org/show_bug.cgi?id=794789
3536 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
3538 * gst/rtsp-server/rtsp-client.c:
3539 rtsp-client:Error handling when equal http session cookie
3540 There are some clients that are sending same session cookie on random
3542 https://bugzilla.gnome.org/show_bug.cgi?id=753616
3544 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3546 * gst/rtsp-server/rtsp-media-factory-uri.c:
3547 rtsp-media-factory-uri: Fix compilation with latest GLib
3548 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
3549 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
3550 data->factory = g_object_ref (factory);
3553 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3561 === release 1.14.0 ===
3563 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3569 * gst-rtsp-server.doap:
3573 === release 1.13.91 ===
3575 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
3581 * gst-rtsp-server.doap:
3585 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
3587 * gst/rtsp-server/Makefile.am:
3588 * gst/rtsp-server/meson.build:
3589 * gst/rtsp-server/rtsp-address-pool.h:
3590 * gst/rtsp-server/rtsp-auth.h:
3591 * gst/rtsp-server/rtsp-client.h:
3592 * gst/rtsp-server/rtsp-context.h:
3593 * gst/rtsp-server/rtsp-media-factory-uri.h:
3594 * gst/rtsp-server/rtsp-media-factory.h:
3595 * gst/rtsp-server/rtsp-media.h:
3596 * gst/rtsp-server/rtsp-mount-points.h:
3597 * gst/rtsp-server/rtsp-onvif-client.h:
3598 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3599 * gst/rtsp-server/rtsp-onvif-media.h:
3600 * gst/rtsp-server/rtsp-onvif-server.h:
3601 * gst/rtsp-server/rtsp-params.h:
3602 * gst/rtsp-server/rtsp-permissions.h:
3603 * gst/rtsp-server/rtsp-sdp.h:
3604 * gst/rtsp-server/rtsp-server-prelude.h:
3605 * gst/rtsp-server/rtsp-server.h:
3606 * gst/rtsp-server/rtsp-session-media.h:
3607 * gst/rtsp-server/rtsp-session-pool.h:
3608 * gst/rtsp-server/rtsp-session.h:
3609 * gst/rtsp-server/rtsp-stream-transport.h:
3610 * gst/rtsp-server/rtsp-stream.h:
3611 * gst/rtsp-server/rtsp-thread-pool.h:
3612 * gst/rtsp-server/rtsp-token.h:
3613 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3614 We need different export decorators for the different libs.
3615 For now no actual change though, just rename before the release,
3616 and add prelude headers to define the new decorator to GST_EXPORT.
3618 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3620 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3621 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3622 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3624 === release 1.13.90 ===
3626 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3632 * gst-rtsp-server.doap:
3636 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3638 * gst/rtsp-server/rtsp-media-factory.c:
3639 * gst/rtsp-server/rtsp-permissions.c:
3640 permissions: add Since tags and example for new API
3642 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3644 * docs/libs/gst-rtsp-server-sections.txt:
3645 * gst/rtsp-server/rtsp-media-factory.c:
3646 * gst/rtsp-server/rtsp-media-factory.h:
3647 * gst/rtsp-server/rtsp-permissions.c:
3648 * gst/rtsp-server/rtsp-permissions.h:
3649 * tests/check/gst/permissions.c:
3650 permissions: more bindings-friendly API
3651 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3653 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3656 meson: enable more warnings
3658 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3660 * gst/rtsp-server/rtsp-client.c:
3661 rtsp-client: Place netaddress meta on packets received via TCP
3662 This allows us to later map signals from rtpbin/rtpsource back to the
3663 corresponding stream transport, and allows to do keep-alive based on
3664 RTCP packets in case of TCP media transport.
3665 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3667 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3669 * gst/rtsp-sink/gstrtspclientsink.c:
3670 rtspclientsink: if OPEN failed, unqueue next command
3671 As READY_TO_PAUSED can no longer return async, the RECORD
3672 command will be queued before the OPEN command fails
3673 (for example in case the server could not be connected),
3674 and record then waits for ever.
3675 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3677 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3679 * gst/rtsp-sink/gstrtspclientsink.c:
3680 rtspclientsink: fix retrieval of custom payloader caps
3681 If a bin is passed as the custom payloader, the caps of
3682 its factory will be empty, the correct way to obtain the caps
3683 is to query its sinkpad.
3685 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3687 * gst/rtsp-sink/gstrtspclientsink.c:
3688 rtspclientsink: fix extra unref of custom payloader
3690 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3692 * gst/rtsp-sink/gstrtspclientsink.c:
3693 rspclientsink: fix recent code indentation
3695 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3697 * gst/rtsp-sink/gstrtspclientsink.c:
3698 rtspclientsink: add missing get_type prototype
3700 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3702 * gst/rtsp-sink/gstrtspclientsink.c:
3703 rtspclientsink: allow setting payloader as pad property
3704 This was a FIXME item, and can be quite useful, also
3705 allowing to specify payloader properties from the command
3706 line, which is always nice.
3707 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3709 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3711 * gst/rtsp-server/rtsp-media.c:
3712 rtsp-media: Replace g_print() log line
3713 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3715 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3717 * gst/rtsp-server/rtsp-media.c:
3718 * tests/check/gst/rtspclientsink.c:
3719 rtsp-media: fix RECORD getting stuck
3720 The test_record case was working because async=false had
3721 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3722 but that was incorrect, as it should not be needed.
3723 Removing async=false made the test fail as expected, this is
3724 fixed by not trying to preroll when preparing the media for
3725 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3726 and our peer will not start sending media until it has received
3727 a response to that request, and sent and received a response
3728 to RECORD as well, thus obviously preventing preroll.
3729 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3731 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3733 * gst/rtsp-server/rtsp-auth.c:
3734 rtsp-auth: fix set_tls_authentication_mode annotation
3736 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3738 * gst/rtsp-server/rtsp-onvif-media.c:
3739 rtp-server: remove redefined variable
3740 res is a boolean variable which is defined in the function scope and
3741 redefined, with no reason, in the loop scope. This patch removes the
3743 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3745 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3747 * gst/rtsp-server/rtsp-media.c:
3748 * gst/rtsp-server/rtsp-stream.c:
3749 * gst/rtsp-server/rtsp-stream.h:
3750 stream: Add functions for checking if stream is receiver or sender
3751 ...and replace all checks for RECORD in GstRTSPMedia which are really
3752 for "sender-only". This way the code becomes more generic and introducing
3753 support for onvif-backchannel later on will require no changes in
3756 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3758 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3759 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3760 onvif: Make requires_backchannel() public
3761 ...in order to let subclasses building the onvif part of the pipeline
3762 check whether backchannel shall be included or not.
3764 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3766 * gst/rtsp-server/rtsp-onvif-media.c:
3767 rtsp-server: Switch around sendonly/recvonly attributes
3768 They are wrong in the ONVIF streaming spec. The backchannel should be
3769 recvonly and the normal media should be sendonly: direction is always
3770 from the point of view of the SDP offerer (the server) according to
3773 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3775 * docs/libs/gst-rtsp-server-docs.sgml:
3776 * docs/libs/gst-rtsp-server-sections.txt:
3777 * examples/.gitignore:
3778 * examples/Makefile.am:
3779 * examples/test-onvif-backchannel.c:
3780 * gst/rtsp-server/Makefile.am:
3781 * gst/rtsp-server/rtsp-media.h:
3782 * gst/rtsp-server/rtsp-onvif-client.c:
3783 * gst/rtsp-server/rtsp-onvif-client.h:
3784 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3785 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3786 * gst/rtsp-server/rtsp-onvif-media.c:
3787 * gst/rtsp-server/rtsp-onvif-media.h:
3788 * gst/rtsp-server/rtsp-onvif-server.c:
3789 * gst/rtsp-server/rtsp-onvif-server.h:
3790 * gst/rtsp-server/rtsp-sdp.c:
3791 * gst/rtsp-server/rtsp-sdp.h:
3792 rtsp: Add support for ONVIF backchannel
3793 This adds a new RTSP server, client, media-factory and media subclass
3794 for handling the specifics of the backchannel. Ideally this later can be
3795 extended with other ONVIF specific features.
3797 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3799 * gst/rtsp-server/rtsp-media.c:
3800 rtsp-media: Add support for sending+receiving medias
3801 We need to add an appsrc/appsink in that case because otherwise the
3802 media bin will be a sink and a source for rtpbin, causing a pipeline
3804 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3806 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3812 === release 1.13.1 ===
3814 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3818 * gst-rtsp-server.doap:
3822 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3824 * gst/rtsp-server/rtsp-session-pool.c:
3825 session-pool: remove nullable return annotation
3826 create_watch can only return NULL from the API guards, no
3829 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3831 * gst/rtsp-server/rtsp-media-factory.c:
3832 * gst/rtsp-server/rtsp-media.c:
3833 set_clock functions: Add nullable annotations
3835 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3837 * gst/rtsp-server/rtsp-auth.c:
3838 * gst/rtsp-server/rtsp-client.c:
3839 * gst/rtsp-server/rtsp-media-factory.c:
3840 * gst/rtsp-server/rtsp-media.c:
3841 * gst/rtsp-server/rtsp-mount-points.c:
3842 * gst/rtsp-server/rtsp-server.c:
3843 * gst/rtsp-server/rtsp-session-media.c:
3844 * gst/rtsp-server/rtsp-session-pool.c:
3845 * gst/rtsp-server/rtsp-session.c:
3846 * gst/rtsp-server/rtsp-stream-transport.c:
3847 * gst/rtsp-server/rtsp-stream.c:
3848 * gst/rtsp-server/rtsp-thread-pool.c:
3849 All around: add annotations and API guards
3851 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3853 * tests/test-cleanup.c:
3854 test-cleanup: bind any port
3855 The meson test suite runs tests in parallel, trying to bind
3856 a single port made the test fail.
3858 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3861 meson: make version numbers ints and fix int/string comparison
3862 WARNING: Trying to compare values of different types (str, int).
3863 The result of this is undefined and will become a hard error
3864 in a future Meson release.
3866 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3868 * gst/rtsp-server/rtsp-context.c:
3869 gst_rtsp_context_get_current: add (skip) annotation
3870 The return value type is defined with G_DEFINE_POINTER_TYPE,
3871 and gi emits the following warning:
3872 Invalid non-constant return of bare structure or union; register as
3873 boxed type or (skip)
3875 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3877 * gst/rtsp-server/rtsp-client.c:
3878 rtsp-client: add type annotations
3879 gi doesn't seem to be able to figure out the type of the
3880 signal parameters when defined with G_DEFINE_POINTER_TYPE
3882 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3885 autotools: use -fno-strict-aliasing where supported
3886 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3888 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3891 meson: use -fno-strict-aliasing where supported
3892 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3894 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3896 * gst/rtsp-server/rtsp-mount-points.c:
3897 mount-points: bail out of loop again when matching mount points
3898 Previous patch led to us iterating the entire sequence. Bail out
3899 of the loop again if we have a match but are moving away from it.
3900 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3902 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3904 * tests/check/gst/mountpoints.c:
3905 tests: mountpoints: add more checks for mount point path matching
3906 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3908 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3910 * gst/rtsp-server/rtsp-mount-points.c:
3911 mount-points: fix matching of paths where there's also an entry with a common prefix
3912 e.g. with the following mount points
3916 _match() would not match /raw/video and /raw/snapshot correctly.
3917 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3919 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3921 * docs/libs/gst-rtsp-server-sections.txt:
3922 * gst/rtsp-server/rtsp-permissions.c:
3923 * gst/rtsp-server/rtsp-permissions.h:
3924 * tests/check/gst/permissions.c:
3925 permissions: add some new API to make this usable from bindings
3926 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3928 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3930 * gst/rtsp-server/rtsp-token.c:
3931 rtsp-token: annotate constructors for bindings
3932 This maps _new_empty() to _new(), which also makes RTSPToken()
3933 work properly now. Since this API wasn't usable from bindings
3934 before, this should hopefully be fine.
3935 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3937 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3939 * docs/libs/gst-rtsp-server-sections.txt:
3940 * gst/rtsp-server/rtsp-token.c:
3941 * gst/rtsp-server/rtsp-token.h:
3942 * tests/check/gst/token.c:
3943 rtsp-token: add some API to set fields from bindings
3944 The existing functions are all vararg-based and as such
3945 not usable from bindings.
3946 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3948 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3950 * tests/check/gst/rtspclientsink.c:
3951 * tests/check/gst/rtspserver.c:
3952 * tests/check/gst/sessionpool.c:
3953 * tests/check/gst/stream.c:
3954 tests: fix indentation
3957 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3959 * tests/check/gst/rtspserver.c:
3960 tests: rtspserver: fix another ref leak
3961 Even if this didn't show up in valgrind.
3963 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3965 * tests/check/gst/rtspclientsink.c:
3966 tests: rtspclientsink: fix leak
3968 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3970 * tests/check/gst/rtspserver.c:
3971 test: rtspserver: plug memory leak in test_no_session_timeout
3972 In test_no_session_timeout, unref the rtsp session object when the
3974 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3976 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3978 * gst/rtsp-sink/gstrtspclientsink.c:
3979 rtpsclientsink: Initialize and clear newly added mutex and cond
3980 While it *did* work, glib would automatically create new mutex and cond
3981 ... which never got freed
3983 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3985 * gst/rtsp-server/rtsp-stream.c:
3986 rtsp-stream: Set multicast TTL on the multicast sockets
3987 And not if we do unicast UDP.
3988 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3990 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3992 * gst/rtsp-server/rtsp-stream.c:
3993 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3994 In the multicast case (as in test-multicast, not test-multicast2), the
3995 address could be allocated/reserved (and thus set) already without
3996 allocating the actual socket. We need to allocate the socket here still
3997 instead of just claiming that it was already allocated.
3998 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
4000 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4002 * gst/rtsp-sink/gstrtspclientsink.c:
4003 * gst/rtsp-sink/gstrtspclientsink.h:
4004 rtspclientsink: Use the new rtsp-stream API
4005 https://bugzilla.gnome.org/show_bug.cgi?id=790412
4007 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4009 * gst/rtsp-sink/gstrtspclientsink.c:
4010 * gst/rtsp-sink/gstrtspclientsink.h:
4011 rtspclientsink: Wait until OPEN has been scheduled
4012 Make sure that the sink thread has started opening connection
4013 to the server before continuing.
4014 https://bugzilla.gnome.org/show_bug.cgi?id=790412
4016 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
4019 Automatic update of common submodule
4020 From e8c7a71 to 3fa2c9e
4022 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
4024 * gst/rtsp-server/rtsp-media.c:
4025 * gst/rtsp-server/rtsp-session-media.c:
4026 * gst/rtsp-server/rtsp-stream.c:
4027 rtsp-server: Minor doc fixes
4030 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4033 * tests/Makefile.am:
4034 tests: disable all tests when --disable-tests is used
4035 Move conditional subdir include into top level.
4036 Based on patch by: Joel Holdsworth
4037 https://bugzilla.gnome.org/show_bug.cgi?id=757703
4039 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
4042 * meson_options.txt:
4043 * tests/meson.build:
4044 meson: build more tests and add options to disable tests and examples
4046 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
4048 * gst/rtsp-server/rtsp-session.c:
4049 Fix build when -Werror=deprecated-declarations is on
4050 As gst_rtsp_session_next_timeout is deprecated.
4052 ../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]
4053 res = (gst_rtsp_session_next_timeout (session, now) == 0);
4055 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
4056 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
4057 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
4060 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
4063 Automatic update of common submodule
4064 From 3f4aa96 to e8c7a71
4066 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4068 * tests/check/gst/media.c:
4069 check/media: Add seekability test case: not all streams are active
4070 Media contains two streams but only one is complete and prepared
4072 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4074 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4076 * gst/rtsp-server/rtsp-stream.c:
4077 rtsp-stream: Do not reset 'blocking' if stream is already blocked
4078 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4080 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
4082 * gst/rtsp-server/rtsp-media.c:
4083 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
4084 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4086 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
4089 meson: remove vs_module_defs_dir variable which is no longer needed
4091 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
4093 * gst/rtsp-server/rtsp-session.h:
4096 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
4099 * gst/rtsp-server/meson.build:
4101 * win32/common/libgstrtspserver.def:
4102 win32: remove .def file with exports
4103 They're no longer needed, symbol exporting is now explicit
4104 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
4106 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4109 autotools: stop controlling symbol visibility with -export-symbols-regex
4110 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
4111 This should result in consistent behaviour for the autotools and
4114 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4116 * gst/rtsp-server/rtsp-media.h:
4117 * gst/rtsp-server/rtsp-server.h:
4118 * gst/rtsp-server/rtsp-session.c:
4119 * gst/rtsp-server/rtsp-session.h:
4120 rtsp-server: add missing GST_EXPORT and export deprecated funcs
4122 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
4124 * tests/check/gst/media.c:
4125 check: Add seekability testing on medias
4126 Make sure that once GstRTSPMedia are prepared they returned
4127 the expected seekability results
4128 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4130 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
4132 * docs/libs/gst-rtsp-server-sections.txt:
4133 * gst/rtsp-server/rtsp-media.c:
4134 * gst/rtsp-server/rtsp-stream.c:
4135 * gst/rtsp-server/rtsp-stream.h:
4136 * win32/common/libgstrtspserver.def:
4137 rtsp-media: Enable seeking query before pipeline is complete
4138 SDP are now provided *before* the pipeline is fully complete. In order
4139 to know whether a media is seekable or not therefore requires asking
4140 the invididual streams.
4141 API: gst_rtsp_stream_seekable
4142 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4144 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
4146 * gst/rtsp-server/rtsp-media.c:
4147 rtsp-media: Fix handling in default_unsuspend()
4148 Handle the case when streams are not blocked and media
4149 is suspended from PAUSED.
4150 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
4151 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4153 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
4155 * tests/check/gst/media.c:
4156 check/media: Fix thread pool leak.
4157 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
4158 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4160 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
4162 * gst/rtsp-server/rtsp-media.c:
4163 rtsp-media: Removed fakesink elements
4164 There is not need of adding fakesink elements to the media
4165 pipeline in the dynamic-payloader case.
4166 The media pipeline itself is dynamically updated with
4167 the receiver and sender parts that are based on the client
4168 transport information known after SETUP has been received.
4169 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
4170 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4172 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
4174 * gst/rtsp-server/rtsp-media.c:
4175 rtsp-media: Corrected ASYNC_DONE handling
4176 Media is complete when all the transport based parts are
4177 added to the media pipeline. At this point ASYNC_DONE is
4178 posted by the media pipeline and media is ready to enter
4180 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
4181 https://bugzilla.gnome.org/show_bug.cgi?id=790674
4183 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
4185 * tests/check/gst/media.c:
4186 check/media: Check that prepared media can provide a SDP
4187 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
4189 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
4191 * gst/rtsp-server/rtsp-client.c:
4192 rtsp-client: Don't leak addr
4195 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
4197 * gst/rtsp-server/rtsp-client.c:
4198 * gst/rtsp-server/rtsp-session-media.c:
4199 * gst/rtsp-server/rtsp-stream.c:
4202 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
4204 * gst/rtsp-server/rtsp-media.c:
4205 rtsp-media: Don't unblock with remaining dynamic payloaders
4206 If we still have some dynamic paylaoders which haven't posted
4207 no-more-pads yet, don't go to PREPARED if one of the streams
4209 The risk was that we would end up not exposing/using all specified
4211 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
4212 then it will take a bit more time to start. But only if those 3
4213 conditions are present.
4214 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4216 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
4218 * gst/rtsp-server/rtsp-media.c:
4221 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
4223 * gst/rtsp-server/rtsp-media.c:
4224 rtsp-media: Don't set float on a gint64 variable
4225 Just use 0. Fixes 'undefined' behaviour from clang
4227 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
4229 * gst/rtsp-server/rtsp-media.c:
4230 rtsp-media: Fix previous commit
4231 We only want to count dynamic payloaders
4233 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
4235 * gst/rtsp-server/rtsp-media.c:
4236 * tests/check/gst/media.c:
4237 rtsp-media: Handle multiple dynamic elements
4238 If we have more than one dynamic payloader in the pipeline, we need
4239 to wait until the *last* one emits 'no-more-pads' before switching
4241 Failure to do so would result in a race where some of the streams
4242 wouldn't properly be prepared
4243 https://bugzilla.gnome.org/show_bug.cgi?id=769521
4245 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4247 * win32/common/libgstrtspserver.def:
4248 win32: Fix exported symbols list
4250 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
4252 * gst/rtsp-server/rtsp-stream.c:
4253 rtsp-stream: Only update the RTP udpsink if it actually exists
4254 For send-only streams it does not exist, but the RTCP udpsink might.
4256 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
4258 * win32/common/libgstrtspserver.def:
4259 win32: Update exports
4261 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
4263 * gst/rtsp-server/rtsp-media.c:
4264 * gst/rtsp-server/rtsp-stream.c:
4265 * gst/rtsp-server/rtsp-stream.h:
4266 rtsp-media: seek on media pipelines that are complete
4267 Make sure that a seek is performed on pipelines that
4268 contain at least one sink element.
4269 Change-Id: Icf398e10add3191d104b1289de612412da326819
4270 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4272 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
4274 * gst/rtsp-server/rtsp-client.c:
4275 * gst/rtsp-server/rtsp-media.c:
4276 * gst/rtsp-server/rtsp-media.h:
4277 * gst/rtsp-server/rtsp-stream.c:
4278 * gst/rtsp-server/rtsp-stream.h:
4279 * tests/check/gst/client.c:
4280 * tests/check/gst/media.c:
4281 * tests/check/gst/rtspserver.c:
4282 * tests/check/gst/stream.c:
4283 Dynamically reconfigure pipeline in PLAY based on transports
4284 The initial pipeline does not contain specific transport
4285 elements. The receiver and the sender parts are added
4287 If the media is shared, the streams are dynamically
4288 reconfigured after each PLAY.
4289 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4291 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
4293 * gst/rtsp-server/rtsp-stream.c:
4294 rtsp-stream: obtain stream position from pad
4295 If no sinks have been added yet, obtain the current and
4296 the stop position of the stream from the send_src pad.
4297 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
4298 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4300 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
4302 * gst/rtsp-server/rtsp-session-media.c:
4303 * gst/rtsp-server/rtsp-session-media.h:
4304 rtsp-session-media: add function to get a list of transports
4305 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
4306 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4308 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
4310 * gst/rtsp-server/rtsp-stream.c:
4311 * gst/rtsp-server/rtsp-stream.h:
4312 rtsp-stream: add functions to get rtp and rtcp multicast sockets
4313 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
4314 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4316 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
4318 * gst/rtsp-server/rtsp-stream.c:
4319 stream: set async=sync=false only for RTCP appsink
4320 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
4321 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4323 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
4325 * gst/rtsp-server/rtsp-media.c:
4326 rtsp-media: return minimum value in query position case
4327 The minimum position should be returned as we are interested
4328 in the whole interval.
4329 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
4330 https://bugzilla.gnome.org/show_bug.cgi?id=788340
4332 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
4334 * gst/rtsp-server/rtsp-session.c:
4335 * tests/check/gst/rtspserver.c:
4336 rtsp-session: Handle the case when timeout=0
4337 According to the documentation, a timeout of value 0 means
4338 that the session never timeouts. This adds handling of that.
4339 If timeout=0 we just return with a -1 from
4340 gst_rtsp_session_next_timeout_usec ().
4341 https://bugzilla.gnome.org/show_bug.cgi?id=785058
4343 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4345 * gst/rtsp-sink/gstrtspclientsink.c:
4346 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
4347 https://bugzilla.gnome.org/show_bug.cgi?id=785024
4349 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
4351 * docs/libs/gst-rtsp-server-sections.txt:
4352 * gst/rtsp-server/rtsp-media-factory.c:
4353 docs: add media factory transport mode accessors
4354 and fix the documentation for the return value of the getter
4356 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
4358 * gst/rtsp-server/rtsp-client.c:
4359 rtsp-client: unref 'pipelined_requests' in finalize
4360 The hash table priv->pipelined_requests is not unref:ed in the
4361 finalize funktion. Make sure it is.
4362 https://bugzilla.gnome.org/show_bug.cgi?id=788704
4364 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
4366 * gst/rtsp-server/rtsp-media.c:
4367 rtsp-media: Initialize scalar variable
4370 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
4372 * win32/common/libgstrtspserver.def:
4373 win32: Update export file
4375 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4377 * gst/rtsp-server/rtsp-client.c:
4378 * gst/rtsp-server/rtsp-media.c:
4379 * gst/rtsp-server/rtsp-media.h:
4380 Start support for RTSP 2.0
4381 This adds basic support for new 2.0 features, though the protocol is
4382 subposdely backward incompatible, most semantics are the sames.
4385 * version negotiation
4386 * pipelined requests support
4387 * Media-Properties support
4388 * Accept-Ranges support
4390 * gst_rtsp_media_seekable
4391 The RTSP methods that have been removed when using 2.0 now return
4393 https://bugzilla.gnome.org/show_bug.cgi?id=781446
4395 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4397 * gst/rtsp-server/rtsp-stream.c:
4398 stream: Use stream duration as stream-stop if segment was not configured with a stop
4399 Allowing client to know stream duration when no seeking happened.
4400 https://bugzilla.gnome.org/show_bug.cgi?id=783435
4402 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
4404 * gst/rtsp-server/rtsp-media-factory.c:
4405 rtsp-media-factory: Don't cache any media if NULL was returned as key
4406 The docs already mentioned this, but we actually stored it in the hash
4407 table with key==NULL and leaked its reference forever.
4409 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
4411 * gst/rtsp-sink/gstrtspclientsink.c:
4412 * gst/rtsp-sink/gstrtspclientsink.h:
4413 rtspclientsink: Use a mutex for protecting against concurrent send/receives
4414 This is a simple port of:
4415 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
4416 * c438545dc9e2f14f657bc0ef261fff726449867b
4417 * cd17c71dcea5c9310d21f1347c7520983e5869ac
4418 in gst-plugins-good.
4420 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
4422 * gst/rtsp-server/rtsp-sdp.c:
4423 sdp: fix Memory leak in error case
4424 https://bugzilla.gnome.org/show_bug.cgi?id=787059
4426 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4428 * pkgconfig/meson.build:
4429 meson: don't install -uninstalled.pc file
4430 https://bugzilla.gnome.org/show_bug.cgi?id=786457
4432 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
4435 Automatic update of common submodule
4436 From 48a5d85 to 3f4aa96
4438 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4440 * gst/rtsp-server/rtsp-client.c:
4441 rtsp-client: Fix typo in debug message
4443 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
4446 meson: hide symbols by default unless explicitly exported
4448 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4450 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4451 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
4452 Fixes meson warning about undefined @srcdir@.
4454 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
4456 * tests/meson.build:
4457 meson: skip tests on windows for now
4458 As we do in the other modules. As libgstcheck is currently not
4459 built on windows. Fixes "Fallback variable 'gst_check_dep' in
4460 the subproject 'gstreamer' does not exist"" Meson error.
4462 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
4464 * gst/rtsp-server/rtsp-stream.c:
4465 rtsp-stream: fix connection delay due to wrong assumption on last-sample
4466 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
4467 multiudpsink's last-sample always comes from the payloader. Which
4468 is wrong if auxiliary streams are multiplexed in the same stream.
4469 So check the buffer's ssrc against the caps'ssrc before to use its
4470 seqnum. If not the same ssrc just use the payloader as done prior
4471 the commit above or when there is no last-sample yet.
4472 https://bugzilla.gnome.org/show_bug.cgi?id=784094
4474 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4477 meson: Allow using glib as a subproject
4479 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4482 meson: fix with-package-name option
4483 https://bugzilla.gnome.org/show_bug.cgi?id=784082
4485 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4488 Distribute meson_options.txt
4490 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4493 And config.h.meson is no longer dist either
4495 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
4499 meson: config.h.meson is no longer needed
4501 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4503 * tests/check/meson.build:
4504 * tests/meson.build:
4505 meson: Fix building tests and activate them again
4507 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
4509 * tests/check/meson.build:
4510 meson: Do not use path separator in test names
4511 Avoiding warnings like:
4512 WARNING: Target "elements/audioamplify" has a path separator in its name.
4514 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
4517 * meson_options.txt:
4518 meson: add options to set package name and origin
4519 https://bugzilla.gnome.org/show_bug.cgi?id=782172
4521 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4523 * gst/rtsp-server/rtsp-address-pool.h:
4524 * gst/rtsp-server/rtsp-auth.h:
4525 * gst/rtsp-server/rtsp-client.h:
4526 * gst/rtsp-server/rtsp-context.h:
4527 * gst/rtsp-server/rtsp-media-factory-uri.h:
4528 * gst/rtsp-server/rtsp-media-factory.h:
4529 * gst/rtsp-server/rtsp-media.h:
4530 * gst/rtsp-server/rtsp-mount-points.h:
4531 * gst/rtsp-server/rtsp-params.h:
4532 * gst/rtsp-server/rtsp-permissions.h:
4533 * gst/rtsp-server/rtsp-sdp.h:
4534 * gst/rtsp-server/rtsp-server.h:
4535 * gst/rtsp-server/rtsp-session-media.h:
4536 * gst/rtsp-server/rtsp-session-pool.h:
4537 * gst/rtsp-server/rtsp-session.h:
4538 * gst/rtsp-server/rtsp-stream-transport.h:
4539 * gst/rtsp-server/rtsp-stream.h:
4540 * gst/rtsp-server/rtsp-thread-pool.h:
4541 * gst/rtsp-server/rtsp-token.h:
4542 Mark symbols explicitly for export with GST_EXPORT
4544 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4547 * gst/rtsp-sink/Makefile.am:
4548 Remove plugin specific static build option
4549 Static and dynamic plugins now have the same interface. The standard
4550 --enable-static/--enable-shared toggle are sufficient.
4552 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
4558 === release 1.12.0 ===
4560 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
4566 * gst-rtsp-server.doap:
4570 === release 1.11.91 ===
4572 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
4578 * gst-rtsp-server.doap:
4582 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
4585 Automatic update of common submodule
4586 From 60aeef6 to 48a5d85
4588 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4590 * gst/rtsp-server/rtsp-media-factory.c:
4591 * gst/rtsp-server/rtsp-media.c:
4592 * gst/rtsp-server/rtsp-session.c:
4593 * gst/rtsp-server/rtsp-stream.c:
4594 gi: Fix some annotations and docstrings
4596 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4598 * gst/rtsp-server/meson.build:
4600 * meson_options.txt:
4603 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4607 Automatic update of common submodule
4608 From 39ac2f5 to 60aeef6
4610 === release 1.11.90 ===
4612 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4618 * gst-rtsp-server.doap:
4622 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4624 * examples/test-launch.c:
4625 examples: make test-launch pipeline shared by default as well
4627 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4629 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4630 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4631 Just the build dir is not going to work for srcdir!=builddir.
4633 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4636 meson: Update version
4638 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4643 === release 1.11.2 ===
4645 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4651 * gst-rtsp-server.doap:
4654 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4657 meson: dist meson build files
4658 Ship meson build files in tarballs, so people who use tarballs
4659 in their builds can start playing with meson already.
4661 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4663 * examples/test-record.c:
4664 examples/test-record: Add extra line to initial printout
4665 Add an example line of how to deliver a stream to the
4668 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4670 * gst/rtsp-server/rtsp-client.c:
4671 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4672 If there is no Content-Length header, no body would be allocated and the
4673 '\0' would also not be appended to the body.
4675 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4677 * gst/rtsp-server/rtsp-client.c:
4678 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4679 While they logically have 0 bytes length, GstRTSPConnection is appending
4680 a '\0' to everything making the size be 1 instead.
4682 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4687 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4689 * gst/rtsp-server/rtsp-session.c:
4690 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4691 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4694 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4699 === release 1.11.1 ===
4701 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4707 * gst-rtsp-server.doap:
4708 * win32/common/libgstrtspserver.def:
4711 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4713 * gst/rtsp-server/rtsp-stream.c:
4714 rtsp-stream: corrected if-statement in _get_server_port()
4715 This bug was accidentally introduced while fixing a segfault
4716 in _get_server_port() function.
4717 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4719 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4721 * gst/rtsp-server/rtsp-stream.c:
4722 * tests/check/gst/stream.c:
4723 rtsp-stream: fixed segmenation fault in _get_server_port()
4724 Calling function gst_rtsp_stream_get_server_port() results in
4725 segmenation fault in the RTP/RTSP/TCP case.
4726 Port that the server will use to receive RTCP makes only
4727 sense in the UDP case, however the function should handle
4728 the TCP case in a nicer way.
4729 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4731 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4733 * gst/rtsp-server/rtsp-media-factory.c:
4734 dosc: Fix a little typo
4735 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4737 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4739 * pkgconfig/Makefile.am:
4740 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4741 * pkgconfig/meson.build:
4742 meson: generate pkg-config -uninstalled pc files
4743 Generating those files is useful for users building the GStreamer stack
4744 using meson and having to link it to another project which is still
4745 using the autotools.
4746 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4748 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4750 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4751 pkgconfig: fix -uninstalled pc file
4752 pcfiledir was never defined so the paths were wrong.
4753 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4755 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4757 * gst/rtsp-server/rtsp-stream.c:
4758 * tests/check/gst/rtspserver.c:
4759 rtsp-stream: Fixed TCP transport case
4760 Make sure that the appsink element is actually added to
4761 the bin before trying to link it with the elements in it.
4762 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4764 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4770 Remove generated .spec file
4771 Likely extremely bitrotten, and we should not ship this anyway.
4773 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4776 Automatic update of common submodule
4777 From f980fd9 to 39ac2f5
4779 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4781 * gst/rtsp-server/rtsp-media.c:
4782 media: Fix pt map caps
4783 Since decryption is handled within rtpbin, all outcoming stream
4784 caps will be application/x-rtp (i.e. regular rtp)
4785 Fixes RECORD with SRTP streams
4787 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4789 * gst/rtsp-server/rtsp-media-factory.c:
4790 media-factory: Create media objects with the proper transport mode
4791 The function called immediately afterwards (collect_streams()) will
4792 need it to work properly
4794 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4796 * gst/rtsp-server/rtsp-auth.c:
4797 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4799 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4801 * gst/rtsp-server/rtsp-media-factory.c:
4802 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4803 We're going to put a pipeline into a pipeline otherwise, which is not
4806 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4808 * gst/rtsp-server/rtsp-media.c:
4809 media: Fix race condition around finish_unprepare() if called multiple time
4810 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4812 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4814 * gst/rtsp-sink/gstrtspclientsink.c:
4815 rtspclientsink: Don't leave stale pointer after unref
4816 Fix a warning on shutdown - don't keep a pointer to an
4817 alread-unreffed object.
4819 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4822 common: use https protocol for common submodule
4823 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4825 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4827 * gst/rtsp-server/rtsp-stream.c:
4828 stream: block the output of rtpbin instead of the source pipeline
4829 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4830 detection of the srtp rollover counter to add to the SDP.
4831 Unfortunately, it was incomplete for live pipelines where the logic
4832 blocks the source bin before creating the SDP and thus would never have
4833 the necessary informaiton to create a correct SDP with srtp encryption.
4834 Move the pad blocks to rtpbin's output pads instead so that the
4835 necessary information can be created before we need the information for
4837 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4839 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4841 * gst/rtsp-server/rtsp-client.c:
4842 rtsp-client: add IDLE timeout, before session exists
4843 The RTSP server will not timeout an idle RTSP connection
4844 (note this is different from doing timeout on a RTSP
4846 At least for Apache this is a problem when running RTSP over
4847 HTTPS since it uses one of the threads (there is a rather
4848 limited number) that are available for handling requests.
4849 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4851 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4856 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4858 * gst/rtsp-server/rtsp-stream.c:
4859 rtsp-stream: Set close-socket FALSE on UDP src:es
4860 With this RTSP server can use the sockets independent on the udpsrc
4862 When the udp src is finalized it will unref socket and when g_socket
4863 is finalized the socket will be closed.
4864 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4866 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4868 * gst/rtsp-sink/gstrtspclientsink.c:
4869 rtspclientsink: Move to new helper function to parse authentication responses
4870 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4872 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4874 * examples/Makefile.am:
4875 * examples/test-auth-digest.c:
4876 * gst/rtsp-server/rtsp-auth.c:
4877 * gst/rtsp-server/rtsp-auth.h:
4878 * win32/common/libgstrtspserver.def:
4879 rtsp-auth: Add support for Digest authentication
4880 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4882 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4885 * gst/rtsp-server/meson.build:
4887 * tests/check/meson.build:
4889 * win32/common/libgstrtspserver.def:
4890 Enable building with MSVC
4891 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4893 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4896 meson: gstreamer gst_check_dep does not exist on windows
4898 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4900 * gst/rtsp-server/rtsp-client.c:
4901 client: update do_send_message to match type GstRTSPClientSendFunc
4902 This type mismatch fails building with MSVC
4903 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4905 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4907 * gst/rtsp-server/rtsp-sdp.c:
4908 rtsp-sdp: Fix indentation
4910 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4912 * gst/rtsp-server/rtsp-media.c:
4913 rtsp-media: Only signal "new-state" if the state has actually changed
4914 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4916 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4918 * gst/rtsp-server/rtsp-client.c:
4919 * gst/rtsp-server/rtsp-client.h:
4920 client: emit signal in the beginning of each rtsp request
4921 These signals let the application validate the requests, configure the
4922 media/stream in a certain way and also generate error status code in
4923 case of error or bad request.
4924 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4926 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4929 meson: update version
4931 === release 1.11.0 ===
4933 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4938 === release 1.10.0 ===
4940 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4946 * gst-rtsp-server.doap:
4949 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4951 * tests/check/gst/rtspserver.c:
4952 * tests/check/gst/stream.c:
4953 tests: try to avoid using the same ports in different tests
4954 Causes problems with client multicast tests otherwise if
4955 tests are run in parallel.
4956 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4958 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4960 * tests/check/gst/client.c:
4961 tests: client: use fail_unless_equals_foo() for better failure reporting
4963 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4965 * gst/rtsp-server/rtsp-client.c:
4966 rtsp-client: Session filter in unwatch session
4967 Call session filter with filter_session_media as paramer in
4968 client_unwatch_session if using drop_backlog = FALSE.
4969 In client_unwatch_session its allowed to grow the watchs backlog.
4970 If using drop_backlog = FALSE and the backlog is full it will cause
4971 a deadlock when setting session media state to NULL
4972 if the backlog is not allowed to grow.
4973 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4975 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4978 meson: add fallbacks for gst modules
4981 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4983 * gst/rtsp-server/rtsp-client.c:
4984 rtsp-client: Fix factory leaking in find_media() in error cases
4985 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4987 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4989 * gst/rtsp-server/rtsp-stream.c:
4990 stream: Fix randomly missing streams from SDP with dynamic elements
4991 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4992 "pad-added" signal. In that case priv->srcpad could already have its caps,
4993 and they'll be sent to priv->send_src[0] pad. That means that when it
4994 connects "notify::caps" signal, that pad could already have received its
4995 caps and the signal won't be emitted anymore.
4996 In that case priv->caps stay to NULL and when building the SDP that stream
4997 gets ignored. Leading to missing video or audio when playing in client side.
4998 https://bugzilla.gnome.org/show_bug.cgi?id=772478
5000 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
5003 meson: update version
5005 === release 1.9.90 ===
5007 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5013 * gst-rtsp-server.doap:
5016 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
5018 * gst/rtsp-server/rtsp-media-factory.c:
5019 * gst/rtsp-server/rtsp-media.c:
5020 * gst/rtsp-server/rtsp-stream.c:
5021 rtsp-server: Hint that set_multicast_iface expects the name of the interface
5022 To prevent any possibly confusion with IPs or anything else.
5023 https://bugzilla.gnome.org/show_bug.cgi?id=771530
5025 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
5027 * gst/rtsp-server/rtsp-media-factory.c:
5028 * gst/rtsp-server/rtsp-media.c:
5029 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
5030 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
5032 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
5035 configure: Depend on gstreamer 1.9.2.1
5037 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
5041 Automatic update of common submodule
5042 From b18d820 to f980fd9
5044 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
5048 Automatic update of common submodule
5049 From 6f2d209 to b18d820
5051 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
5053 * gst/rtsp-server/rtsp-stream.c:
5054 rtsp-stream: Remove unused _locked() variant of a function
5055 It was added during refactoring.
5057 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5059 * gst/rtsp-server/rtsp-stream.c:
5060 stream: cosmetic cleanup
5061 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5063 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5065 * gst/rtsp-server/rtsp-stream.c:
5066 stream: Compare IP addresses case insensitive in more places
5067 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5069 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5072 * gst/rtsp-server/rtsp-stream.c:
5073 stream: Fix leaked joined_bin
5074 There is no need to keep a strong ref on it, and _leave_bin() was
5075 setting it to NULL before calling g_clear_object() so it was leaked.
5076 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5078 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5080 * gst/rtsp-server/rtsp-stream.c:
5081 rtsp-stream: Compare IP address strings case insensitive
5082 Otherwise IPv6 addresses might fail this comparision.
5084 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
5086 * gst/rtsp-server/rtsp-stream.c:
5087 rtsp-stream: Bind multicast sockets to ANY as before
5088 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
5090 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
5092 * gst/rtsp-server/rtsp-session.c:
5093 rtsp-session: Fix segfault when doing keep-alive after removing the session
5094 If keep-alive happens after removing the session but before finalizing the
5095 stream transport, we would segfault.
5096 https://bugzilla.gnome.org/show_bug.cgi?id=750544
5098 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
5100 * gst/rtsp-server/rtsp-stream.c:
5101 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
5102 Adding them later will cause deadlocks due to
5103 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
5104 2) adding the multicast sink
5105 3) waiting for it to get data to preroll again
5106 3) never happens because the queues after the tee are full.
5108 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
5110 * gst/rtsp-server/rtsp-stream.c:
5111 rtsp-stream: Fix up various multicast related issues
5113 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
5115 * tests/check/gst/stream.c:
5116 tests: Fix compilation
5118 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5120 * gst/rtsp-server/rtsp-client.c:
5121 * gst/rtsp-server/rtsp-stream.c:
5122 * tests/check/gst/stream.c:
5123 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
5124 This is basically reverting changes introduced in commit f62a9a7,
5125 because it was introducing various regressions:
5126 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
5127 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
5128 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
5129 - If a mcast client connects, it creates a new socket in SETUP to try to respect
5130 the destination/port given by the client in the transport, and overrides the
5131 socket already set on the udpsink element. That means that if we already had a
5132 client connected, the source address on the udp packets it receives suddenly
5134 - If a 2nd mcast client connects, the destination/port in its transport is
5135 ignored but its transport wasn't updated.
5136 What this patch does:
5137 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
5138 - Always have a tee+queue when udp is enabled. This could be optimized
5139 again in a later patch, but is more complicated. If no unicast clients
5140 connects then those elements are useless, this could be also optimized
5142 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
5143 seperated from those for unicast clients. Since we already support only
5144 one mcast address, we also create only one set of elements.
5145 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5147 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5149 * gst/rtsp-server/rtsp-stream.c:
5150 stream: factor our plug_src function
5151 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5153 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5155 * gst/rtsp-server/rtsp-stream.c:
5156 stream: factor out plug_sink function
5157 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5159 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5161 * gst/rtsp-server/rtsp-stream.c:
5162 stream: small documentation clarification
5163 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5165 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5167 * gst/rtsp-server/rtsp-stream.c:
5168 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
5169 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5171 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5173 * gst/rtsp-server/rtsp-stream.c:
5174 stream: Keep a ref on joined bin
5175 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5177 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5179 * gst/rtsp-server/rtsp-stream.c:
5180 stream: code cleanup
5181 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5183 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5185 * gst/rtsp-server/rtsp-stream.c:
5186 stream: small fix in error code path
5187 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5189 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5191 * gst/rtsp-server/rtsp-stream.c:
5192 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
5193 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
5194 but keeps unit tests.
5195 https://bugzilla.gnome.org/show_bug.cgi?id=766612
5197 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
5202 === release 1.9.2 ===
5204 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
5210 * gst-rtsp-server.doap:
5213 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
5216 * examples/meson.build:
5218 * gst/rtsp-server/meson.build:
5219 * gst/rtsp-sink/meson.build:
5221 * pkgconfig/meson.build:
5222 * tests/check/meson.build:
5223 * tests/meson.build:
5224 Add support for Meson as alternative/parallel build system
5225 https://github.com/mesonbuild/meson
5227 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
5230 * tests/check/Makefile.am:
5231 build: silence error about pthread for 'make check' in osx
5232 Fixes "clang: error: argument unused during compilation: '-pthread'"
5234 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
5236 * gst/rtsp-server/rtsp-client.c:
5237 rtsp-client: Fix leaking of media in error cases
5238 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
5239 and myself to make the media refcounting a bit easier to follow.
5240 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5242 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
5244 * gst/rtsp-server/rtsp-client.c:
5245 rtsp-client: Fix leaking of session in error cases
5246 https://bugzilla.gnome.org/show_bug.cgi?id=755632
5248 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
5251 Automatic update of common submodule
5252 From f363b32 to f49c55e
5254 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
5259 === release 1.9.1 ===
5261 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
5267 * gst-rtsp-server.doap:
5270 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
5273 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
5274 https://bugzilla.gnome.org/show_bug.cgi?id=767463
5276 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5279 Automatic update of common submodule
5280 From ac2f647 to f363b32
5282 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5284 * gst/rtsp-server/rtsp-sdp.c:
5285 * gst/rtsp-server/rtsp-sdp.h:
5286 * gst/rtsp-server/rtsp-stream.c:
5287 * gst/rtsp-server/rtsp-stream.h:
5288 sdp: add rollover counters for all sender SSRC
5289 We add different crypto sessions in MIKEY, one for each sender
5290 SSRC. Currently, all of them will have the same security policy, 0.
5291 The rollover counters are obtained from the srtpenc element using the
5293 https://bugzilla.gnome.org/show_bug.cgi?id=730539
5295 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5297 * gst/rtsp-server/rtsp-media-factory.h:
5298 * gst/rtsp-server/rtsp-server.h:
5299 docs: fix some typos
5301 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
5303 * gst/rtsp-server/Makefile.am:
5304 g-i: pass compiler env to g-ir-scanner
5305 It's what introspection.mak does as well. Should
5306 fix spurious build failures on gnome-continuous
5307 (caused by g-ir-scanner getting compiler details
5308 via python which is broken in some environments
5309 so passing the compiler details bypasses that).
5311 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
5313 * gst/rtsp-server/rtsp-session.c:
5314 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
5315 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
5316 https://bugzilla.gnome.org/show_bug.cgi?id=766619
5318 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
5320 * gst/rtsp-sink/gstrtspclientsink.c:
5321 rtspclientsink: Check return value of sscanf
5322 And just make sure we always have 0/0 if we have an error
5325 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
5327 * gst/rtsp-server/rtsp-stream.c:
5328 * tests/check/gst/rtspserver.c:
5329 * tests/check/gst/stream.c:
5330 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
5331 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
5332 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
5333 - Create unit test for shared media.
5334 https://bugzilla.gnome.org/show_bug.cgi?id=764744
5336 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5338 * gst/rtsp-server/rtsp-stream.c:
5339 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
5340 For IPv6 addresses, binding to a multicast group does not work on Linux
5341 either. Always bind to ANY and then later join the multicast group.
5342 https://bugzilla.gnome.org/show_bug.cgi?id=764679
5344 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
5347 Automatic update of common submodule
5348 From 6f2d209 to ac2f647
5350 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
5352 * gst/rtsp-server/rtsp-thread-pool.c:
5353 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
5354 Clarified why it is necessary to add source information to
5355 GstRTSPThreadImpl. See the reported bug in GLib:
5356 https://bugzilla.gnome.org/show_bug.cgi?id=720186
5357 for more information.
5358 https://bugzilla.gnome.org/show_bug.cgi?id=761702
5360 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
5362 * examples/Makefile.am:
5363 examples: Clean up CFLAGS/LDADD even more
5364 The internal .la should come first and is part of LDADD, as is
5367 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
5369 * examples/Makefile.am:
5370 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
5372 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
5374 * gst/rtsp-server/Makefile.am:
5375 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
5377 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5379 * gst/rtsp-server/rtsp-client.c:
5380 * gst/rtsp-server/rtsp-media-factory.c:
5381 * gst/rtsp-server/rtsp-media-factory.h:
5382 * gst/rtsp-server/rtsp-media.c:
5383 * gst/rtsp-server/rtsp-media.h:
5384 * gst/rtsp-server/rtsp-sdp.c:
5385 * gst/rtsp-server/rtsp-stream.c:
5386 * gst/rtsp-server/rtsp-stream.h:
5387 rtsp-server: Implement clock signalling according to RFC7273
5388 For NTP and PTP clocks we signal the actual clock that is used and signal
5389 the direct media clock offset.
5390 For all other clocks we at least signal that it's the local sender clock.
5391 This allows receivers to know which clock was used to generate the media and
5392 its RTP timestamps. Receivers can then implement network synchronization,
5393 either absolute or at least relative by getting the sender clock rate directly
5394 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
5396 https://bugzilla.gnome.org/show_bug.cgi?id=760005
5398 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
5400 * gst/rtsp-sink/gstrtspclientsink.c:
5401 rtspclientsink: Add support for setting the multicast interface
5402 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5404 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5406 * gst/rtsp-server/rtsp-media-factory.c:
5407 * gst/rtsp-server/rtsp-media-factory.h:
5408 * gst/rtsp-server/rtsp-media.c:
5409 * gst/rtsp-server/rtsp-media.h:
5410 * gst/rtsp-server/rtsp-stream.c:
5411 * gst/rtsp-server/rtsp-stream.h:
5412 rtsp-media: Add support for setting the multicast interface
5413 https://bugzilla.gnome.org/show_bug.cgi?id=763000
5415 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
5417 * gst/rtsp-sink/gstrtspclientsink.c:
5418 rtspclientsink: use new gst_element_class_add_static_pad_template()
5419 https://bugzilla.gnome.org/show_bug.cgi?id=763196
5421 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5426 === release 1.8.0 ===
5428 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
5434 * gst-rtsp-server.doap:
5437 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
5439 * gst/rtsp-server/rtsp-stream.c:
5440 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
5441 This would get us NO_PREROLL in the bin again and break seeking.
5442 Thanks to Carlos Rafael Giani for helping to debug this!
5443 https://bugzilla.gnome.org/show_bug.cgi?id=740509
5445 === release 1.7.91 ===
5447 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5453 * gst-rtsp-server.doap:
5456 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5458 * gst/rtsp-server/rtsp-stream.c:
5459 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
5460 Without this, RECORD pipelines are broken because
5461 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
5462 added later. Previously it was there earlier and due to NO_PREROLL caused the
5463 pipeline to preroll immediately
5464 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
5465 as the corresponding code previously was only for PLAY pipelines.
5466 https://bugzilla.gnome.org/show_bug.cgi?id=763281
5468 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
5470 * gst/rtsp-server/rtsp-stream.c:
5471 rtsp-stream: Fix typo in the docstring
5472 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
5474 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
5476 * gst/rtsp-server/rtsp-stream.c:
5477 rtsp-stream: Disable multicast loopback for all our sockets
5478 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
5479 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
5480 loopback setting on the socket... while udpsink does which unfortunately has
5481 no effect here on Windows but on Linux.
5482 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5484 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
5486 * tests/check/gst/stream.c:
5487 stream tests: added new tests
5488 Test a case when the address pool only contains multicast addresses
5489 and the client is requesting unicast udp.
5490 Added tests for multicast ports allocation.
5491 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5493 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
5495 * gst/rtsp-server/rtsp-stream.c:
5496 rtsp-stream: Only bind multicast sockets to ANY on Windows
5497 On Linux it is still needed to bind to the multicast address
5498 to filter out random other packets, while on Windows binding
5499 to multicast addresses just fails.
5501 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5503 * gst/rtsp-server/rtsp-stream.c:
5504 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
5505 Otherwise we fail to allocate UDP ports if the pool only contains multicast
5506 addresses, which is something that used to work before. For unicast addresses
5507 if the pool contains none, we just allocate them as if there is no pool at
5509 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5511 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
5513 * gst/rtsp-server/rtsp-client.c:
5514 * gst/rtsp-server/rtsp-stream.c:
5515 rtsp-server: Fix indentation
5517 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5519 * gst/rtsp-server/rtsp-stream.c:
5520 rtsp-stream: Don't bind the sockets to multicast addresses
5521 This works on Linux but fails completely on Windows. You're supposed
5522 to bind to ANY and then join the multicast group.
5523 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5525 === release 1.7.90 ===
5527 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5533 * gst-rtsp-server.doap:
5536 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
5539 Automatic update of common submodule
5540 From b64f03f to 6f2d209
5542 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
5544 * gst/rtsp-sink/gstrtspclientsink.c:
5545 * tests/check/gst/rtspclientsink.c:
5546 rtspsink: Fix some leaks in rtspclientsink and the unit test.
5547 https://bugzilla.gnome.org/show_bug.cgi?id=762525
5549 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
5551 * tests/check/gst/media.c:
5552 * tests/check/gst/rtspclientsink.c:
5553 * tests/check/gst/rtspserver.c:
5554 * tests/check/gst/stream.c:
5555 tests: unit test fixes
5556 Removed port allocation test from the media suite.
5557 The port allocation failure is now in the stream suite.
5559 Make sure that the media is suspended after the DESCRIBE request
5560 before reconfiguring the UDP sinks.
5562 In the RECORD case we have to set async property to false
5563 for the appsink element in the test in order to make sure
5564 that the media pipeline doesn't hang in start_preroll().
5565 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5567 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
5569 * gst/rtsp-server/rtsp-client.c:
5570 * gst/rtsp-server/rtsp-stream.c:
5571 * gst/rtsp-server/rtsp-stream.h:
5572 rtsp-stream: postpone UDP socket allocation until SETUP
5573 Postpone the allocation of the UDP sockets until we know
5574 what transport has been chosen by the client.
5575 Both unicast and multicast UDP sources are created in one
5577 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5579 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
5581 * gst/rtsp-server/rtsp-stream.c:
5582 rtsp-stream: postpone the creation of the UDP sources
5583 Code refactoring: allocate the UDP ports after the sender and
5584 the reciver parts have been created.
5585 We postpone the creation of the UDP sources until the UDP
5586 ports have been allocated.
5587 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5589 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
5591 * gst/rtsp-server/rtsp-stream.c:
5592 rtsp-stream: added function for setting UDP sources to PLAYING state
5593 Code refactoring: Introduced a function for setting UDP sources
5595 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5597 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5599 * gst/rtsp-server/rtsp-stream.c:
5600 rtsp-stream: added function for creating and configuring UDP sources
5601 Code refactoring: create and configure UDP sources in a separate function.
5602 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5604 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5606 * gst/rtsp-server/rtsp-stream.c:
5607 rtsp-stream: added function for RTP/RTCP socket configuration
5608 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5609 in a separate function.
5610 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5612 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5614 * gst/rtsp-server/rtsp-stream.c:
5615 rtsp-stream: added function for creating and configuring UDP sinks
5616 Code refactoring: create and configure UDP sinks in a separate function.
5617 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5619 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5621 * gst/rtsp-server/rtsp-stream.c:
5622 rtsp-stream: added helper function for creating the sender/receiver parts
5623 Code refactoring: introduced helper function for creating
5624 the receiver and the sender parts of the streaming pipeline.
5625 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5627 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5632 === release 1.7.2 ===
5634 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5640 * gst-rtsp-server.doap:
5643 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5645 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5646 uninstalled.pc: add support for non libtool build systems
5647 Currently the .la path is provided which requires to use libtool as
5648 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5649 It is fine as long as the application is built using libtool.
5650 So currently it is not possible to compile a GStreamer application
5651 within gst-uninstalled with CMake or other build system different
5653 This patch allows to do the following in gst-uninstalled env:
5654 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5655 gstreamer-rtsp-server-1.0)
5656 Previously it required to prepend libtool --mode=link
5657 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5659 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5661 * gst/rtsp-sink/gstrtspclientsink.c:
5662 rtspclientsink: remove check for impossible condition
5663 Goto error label checks stream to see if it needs to be unreferenced before
5664 returning, but this goto jumps happens before the stream is ever set, so it
5665 will always be NULL in this error label.
5668 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5670 * gst/rtsp-sink/gstrtspclientsink.c:
5671 rtspclientsink: clean switch statements
5672 Coverity demands for fallthrough statements to be clearly commented,
5673 to distinguish from accidental fall throughs. And it also needs all
5674 cases to finish with a break, even if the break is never going to be
5675 executed like in the case of a continue jump.
5679 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5681 * tests/check/Makefile.am:
5682 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5683 To get the CK_DEFAULT_TIMEOUT defined for all tests
5684 Also removes a 120 seconds timeout that was set as default
5685 explicitly in this module
5686 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5688 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5692 Automatic update of common submodule
5693 From 86e4663 to b64f03f
5695 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5697 * gst/rtsp-server/rtsp-media.c:
5698 rtsp-media: fix state_lock not locked again when preroll fails
5699 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5701 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5704 configure: Move plugin specific flags below all the others
5705 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5706 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5708 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5710 * gst/rtsp-sink/gstrtspclientsink.c:
5711 rtspclientsink: Simplify slightly using new -base API
5712 Use the new Mikey and SDP API in the base plugins libs
5713 to simplify some code.
5714 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5716 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5721 * gst/rtsp-sink/Makefile.am:
5722 * gst/rtsp-sink/gstrtspclientsink.c:
5723 * gst/rtsp-sink/gstrtspclientsink.h:
5724 * gst/rtsp-sink/plugin.c:
5725 * tests/check/Makefile.am:
5726 * tests/check/gst/rtspclientsink.c:
5727 rtspsink: Add rtspclientsink element
5728 Add an rtspclientsink element that accepts streams for which
5729 there is a registered payloader and sends them to
5730 an RTSP server using RECORD.
5731 Sending is synchronised to the pipeline clock. Payload-types
5732 are automatically selected. The 'new-payloader' signal is fired
5733 for custom configuration of payloaders when they are created.
5734 Can now stream a movie like this:
5736 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5737 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5739 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5740 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5741 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5743 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5745 * gst/rtsp-server/rtsp-stream.c:
5746 * gst/rtsp-server/rtsp-stream.h:
5747 rtsp-stream: Add functions for using rtsp-stream from the client
5748 Add a boolean to indicate that the rtsp-stream is running on the
5749 'client' side of an RTSP connection, for sending streams via
5750 RECORD. In that case, the roles of the client/server ports
5751 in transport setup are swapped.
5752 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5754 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5756 * gst/rtsp-server/rtsp-sdp.c:
5757 * gst/rtsp-server/rtsp-sdp.h:
5758 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5759 A new function that adds info from a GstRTSPStream into an SDP message.
5760 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5762 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5764 * gst/rtsp-server/rtsp-media.c:
5765 rtsp-media: Fix mutex beeing unlocked while they should be locked
5766 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5768 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5770 * gst/rtsp-server/rtsp-media-factory.c:
5771 rtsp-media-factory: add missing break in "clock" property setter
5774 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5776 * gst/rtsp-server/rtsp-stream.c:
5777 rtsp-stream: fixed assert during update transport
5778 When RTSP server trying update transport during multicast, it throws an
5779 assert. The assert is thrown because it is trying to get the parent of
5780 an non-existing funnel element.
5781 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5783 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5785 * gst/rtsp-server/rtsp-permissions.h:
5786 * gst/rtsp-server/rtsp-thread-pool.h:
5787 * gst/rtsp-server/rtsp-token.h:
5788 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5789 gtk-doc can handle static inline functions just fine these days,
5790 there's no need for this stuff any more.
5792 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5794 * gst/rtsp-server/rtsp-media.c:
5795 * gst/rtsp-server/rtsp-sdp.c:
5796 sdp: replace duplicated codes to call new base sdp apis
5797 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5799 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5801 * examples/test-netclock.c:
5802 test-netclock: Use the new API to configure a clock directly
5804 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5806 * gst/rtsp-server/rtsp-media-factory.c:
5807 * gst/rtsp-server/rtsp-media-factory.h:
5808 * gst/rtsp-server/rtsp-media.c:
5809 * gst/rtsp-server/rtsp-media.h:
5810 rtsp-media: Add API to directly configure a clock on the media pipelines
5812 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5814 * gst/rtsp-server/rtsp-media.c:
5815 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5817 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5819 * gst/rtsp-server/rtsp-media-factory.c:
5820 rtsp-media-factory: Add FIXME for 2.0
5822 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5824 * gst/rtsp-server/rtsp-stream.c:
5825 rtsp-stream: Fix indentation
5827 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5829 * gst/rtsp-server/rtsp-media.c:
5830 rtsp-media: Do not prepare media after media times out
5831 Deferred calls to start_prepare() can be deferred past the point until
5832 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5833 prepared to wait. Previously there was no lock and no check for this
5834 situation. This meant that a media could be prepared and unprepared
5835 simultaneously by two different threads. Now a lock is in place and a
5836 suitable check is done.
5837 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5839 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5841 * gst/rtsp-server/rtsp-client.c:
5842 * gst/rtsp-server/rtsp-media-factory.c:
5843 * gst/rtsp-server/rtsp-media-factory.h:
5844 * gst/rtsp-server/rtsp-media.c:
5845 * gst/rtsp-server/rtsp-media.h:
5846 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5847 Without TEARDOWN it might be desireable to keep the media running and continue
5848 sending data to the client, even if the RTSP connection itself is
5850 Only do this for session medias that have only UDP transports. If there's at
5851 least on TCP transport, it will stop working and cause problems when the
5852 connection is disconnected.
5853 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5855 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5860 === release 1.7.1 ===
5862 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5868 * gst-rtsp-server.doap:
5871 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5874 configure: Make -Bsymbolic check work with clang.
5875 Update the -Bsymbolic check with the version glib has. This version
5877 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5879 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5881 * gst/rtsp-server/rtsp-session-pool.c:
5882 rtsp-session-pool: Avoid dollar sign ($) in session ids
5883 Live555 in VLC strips off dollar signs and then gets very confused,
5884 we don't loose too much entropy by just skipping it.
5886 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5888 * gst/rtsp-server/rtsp-address-pool.h:
5889 * gst/rtsp-server/rtsp-auth.h:
5890 * gst/rtsp-server/rtsp-client.h:
5891 * gst/rtsp-server/rtsp-media-factory-uri.h:
5892 * gst/rtsp-server/rtsp-media-factory.h:
5893 * gst/rtsp-server/rtsp-media.h:
5894 * gst/rtsp-server/rtsp-mount-points.h:
5895 * gst/rtsp-server/rtsp-permissions.h:
5896 * gst/rtsp-server/rtsp-server.h:
5897 * gst/rtsp-server/rtsp-session-media.h:
5898 * gst/rtsp-server/rtsp-session-pool.h:
5899 * gst/rtsp-server/rtsp-session.h:
5900 * gst/rtsp-server/rtsp-stream-transport.h:
5901 * gst/rtsp-server/rtsp-stream.h:
5902 * gst/rtsp-server/rtsp-thread-pool.h:
5903 * gst/rtsp-server/rtsp-token.h:
5904 rtsp-server: Add g_autoptr() support to all types
5905 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5907 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5909 * gst/rtsp-server/rtsp-stream.c:
5910 rtsp-stream: fixed valgrind error
5911 Fixed the valgrind error in unit test. The UDP source created during
5912 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5914 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5916 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5920 Automatic update of common submodule
5921 From b319909 to 86e4663
5923 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5925 * gst/rtsp-server/rtsp-client.c:
5926 rtsp-client: suspend media during setup request
5927 SETUP request from clients needs to suspend the media to clear the
5928 prerolled buffers. Otherwise it will not affect the prerolled buffer
5929 and the prerolled buffers will be incorrect (for example block-size
5930 from setup request will not affect the prerolled buffer unless the
5931 media is suspended).
5932 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5934 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5936 * gst/rtsp-server/rtsp-stream.c:
5937 rtsp-stream: create stream pipeline based on transport
5938 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5939 only UDP is set as the transport protocol, it will not add the extra tee
5940 or queue element to the pipeline. Both these elements will be added, if
5941 it supports both TCP and UDP protocols. This improves the pipeline
5942 performance when one protocol is present.
5943 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5945 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5947 * gst/rtsp-server/rtsp-stream.c:
5948 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5949 Adding them when not needed will start some logic inside rtpbin that might be
5950 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5951 would start up a rtpjitterbuffer and behave in weird ways.
5952 We still set up the UDP sources for RTP receiving for a sender media to be
5953 able to receive any packets sent by the client for NAT traversal. They will
5954 all go to a fakesink though.
5955 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5956 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5957 receive ASYNC_DONE after a seek.
5958 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5960 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5962 * gst/rtsp-server/rtsp-stream.c:
5963 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5964 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5965 Previously we were only setting this for sender sockets, which caused looped
5966 back packets to be received on Windows if a multicast transport was used.
5968 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5970 * examples/test-record-auth.c:
5971 * examples/test-record.c:
5972 examples: Actually use the provided port in the record examples
5974 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5976 * examples/test-record-auth.c:
5977 test-record-auth: Add the option to build in TLS support
5979 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5981 * examples/test-auth.c:
5982 test-auth: Use an 'anonymous' user for unauthenticated default
5983 There's a comment on one of the resources that 'user' and 'admin'
5984 shouldn't even be able to see it, but they can if the default
5985 token is 'admin2', since that gives them access anyway.
5987 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5989 * examples/.gitignore:
5990 * examples/Makefile.am:
5991 * examples/test-record-auth.c:
5992 Add test-record-auth example
5994 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5996 * gst/rtsp-server/rtsp-client.c:
5997 * tests/check/gst/client.c:
5998 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
6000 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
6002 * gst/rtsp-server/rtsp-server.c:
6003 rtsp-server: Change the logic so we don't pop a NULL context
6004 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
6005 will sometimes fail. This call is made before any context is pushed
6006 resulting in an attempt to pop a NULL context.
6007 https://bugzilla.gnome.org/show_bug.cgi?id=757949
6009 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
6011 * tests/check/gst/rtspserver.c:
6012 rtspserver: Add udp-mcast transport SETUP test
6013 Refactor utility functions in the test file so they can handle
6014 more than UDP and TCP as lower transport.
6015 https://bugzilla.gnome.org/show_bug.cgi?id=756969
6017 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
6019 * gst/rtsp-server/rtsp-stream.c:
6020 rtsp-stream: Always unref return value of gst_object_get_parent()
6021 Fixes a leak of a GstBin in the udp-mcast case.
6022 https://bugzilla.gnome.org/show_bug.cgi?id=756968
6024 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
6027 Automatic update of common submodule
6028 From b99800a to b319909
6030 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
6033 Use new GST_ENABLE_EXTRA_CHECKS #define
6034 https://bugzilla.gnome.org/show_bug.cgi?id=756870
6036 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6039 Automatic update of common submodule
6040 From 6babecd to b99800a
6042 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6045 Update GLib dependency to 2.40.0
6047 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6049 * examples/test-mp4.c:
6050 * gst/rtsp-server/rtsp-stream.c:
6051 stream: listen to sender ssrc signals
6052 https://bugzilla.gnome.org/show_bug.cgi?id=746747
6054 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
6057 common: update for new suppression
6058 Makes check-valgrind pass with glib 2.46
6060 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6062 * gst/rtsp-server/rtsp-media.c:
6063 rtsp-media: Take reference to media that will be prepared
6064 default_prepare() takes a transfer-none reference GstRTSPMedia object.
6065 Later on a g_idle_source_new() is created and a pointer to the media
6066 object is passed as user data. If the media is freed before the idle
6067 source is dispatched the media object pointer is invalid, but the idle
6068 source callback expects it to still be valid. To fix this a reference to
6069 the media object is taken when registering the source callback function
6070 and a corresponding release of the reference is done when the souce is
6072 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
6074 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
6076 * examples/test-launch.c:
6077 * examples/test-mp4.c:
6078 * examples/test-ogg.c:
6079 * examples/test-record.c:
6080 * examples/test-uri.c:
6081 rtsp-server: Fix memory leaks when context parse fails
6082 When g_option_context_parse fails, context and error variables are not getting free'd
6083 which results in memory leaks. Free'ing the same.
6084 And replacing g_error_free with g_clear_error, which checks if the error being passed
6085 is not NULL and sets the variable to NULL on free'ing.
6086 https://bugzilla.gnome.org/show_bug.cgi?id=753863
6088 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
6093 === release 1.6.0 ===
6095 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6101 * gst-rtsp-server.doap:
6104 === release 1.5.91 ===
6106 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
6112 * gst-rtsp-server.doap:
6115 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
6117 * docs/libs/gst-rtsp-server-sections.txt:
6118 * gst/rtsp-server/rtsp-stream.c:
6119 stream: fix docs for recently-added get/set_buffer_size API
6120 https://bugzilla.gnome.org/show_bug.cgi?id=749095
6122 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
6124 * gst/rtsp-server/rtsp-media.c:
6125 rtsp-media: Don't crash on encrypted RTX SDP
6126 In parse_keymgmt(), don't mutate the input string that's been passed
6127 as const, especially since we might need the original value again if
6128 the same key info applies to multiple streams (RTX, for example).
6129 https://bugzilla.gnome.org/show_bug.cgi?id=754753
6131 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
6133 * examples/test-mp4.c:
6134 test-mp4: Support filenames with spaces in them. Error out on too few arguments
6136 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
6138 * examples/test-record.c:
6139 test-record: Check parameter count and print out help
6140 If no launch pipeline was supplied, print out some help
6142 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
6144 * gst/rtsp-server/rtsp-media.c:
6145 * gst/rtsp-server/rtsp-stream.c:
6146 * gst/rtsp-server/rtsp-stream.h:
6147 rtsp-stream: Implement UDP buffer size setting.
6148 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
6150 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
6151 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
6153 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
6155 * gst/rtsp-server/rtsp-media.h:
6156 rtsp-media: Fix small typo causing gtk-doc to complain
6158 === release 1.5.90 ===
6160 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
6166 * gst-rtsp-server.doap:
6169 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6171 * gst/rtsp-server/rtsp-media-factory.c:
6172 media-factory: get port number through gst_rtsp_url_get_port
6173 https://bugzilla.gnome.org/show_bug.cgi?id=753473
6175 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
6177 * tests/check/gst/media.c:
6178 media-test: Removing unnecessary assertion
6179 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6181 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6183 * gst/rtsp-server/rtsp-server.c:
6184 Document that source keeps a ref on server until it's destroyed
6185 https://bugzilla.gnome.org/show_bug.cgi?id=749227
6187 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6189 * tests/check/gst/media.c:
6190 media-test: Test for multiple dynamic payload
6191 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6193 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6195 * gst/rtsp-server/rtsp-media.c:
6196 media: Only add fakesink once per pipeline
6197 The intention is to prevent going PLAYING state before pads are created.
6198 If there was mutilple dynamic payload, it would leak few fakesink and
6199 actually prevent from ever reaching playing state.
6200 https://bugzilla.gnome.org/show_bug.cgi?id=753385
6202 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6204 * gst/rtsp-server/rtsp-media.c:
6205 Revert "rtsp-media: Only add 1 fakesink per pipeline"
6206 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
6208 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6210 * gst/rtsp-server/rtsp-media.c:
6211 rtsp-media: Only add 1 fakesink per pipeline
6212 There should be only one fakesink per pipeline, not per dynpay. This
6213 would lead to element naming clash.
6215 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
6217 * gst/rtsp-server/rtsp-media.c:
6218 rtsp-media: assertion error due to wrong condition check
6219 In media to caps function, reserved_keys array is being used for variable i,
6220 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
6221 changed it to variable j
6222 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6224 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
6226 * gst/rtsp-server/rtsp-media.c:
6227 rtsp-media: Strip keys from the fmtp that we use internally in our caps
6228 Skip keys from the fmtp, which we already use ourselves for the
6229 caps. Some software is adding random things like clock-rate into
6230 the fmtp, and we would otherwise here set a string-typed clock-rate
6231 in the caps... and thus fail to create valid RTP caps
6232 https://bugzilla.gnome.org/show_bug.cgi?id=753009
6234 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6236 * gst/rtsp-server/rtsp-thread-pool.c:
6237 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
6238 https://bugzilla.gnome.org/show_bug.cgi?id=752640
6240 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
6243 Automatic update of common submodule
6244 From f74b2df to 9aed1d7
6246 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
6251 === release 1.5.2 ===
6253 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
6259 * gst-rtsp-server.doap:
6262 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
6264 * gst/rtsp-server/rtsp-client.c:
6265 * gst/rtsp-server/rtsp-client.h:
6266 * tests/check/gst/client.c:
6267 rtsp-client: allow application to decide what requirements are supported
6268 Add "check-requirements" signal and vfunc to allow application
6269 (and subclasses) to check the requirements.
6270 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
6271 https://bugzilla.gnome.org/show_bug.cgi?id=749417
6273 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
6276 Automatic update of common submodule
6277 From 6015d26 to f74b2df
6279 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6281 * gst/rtsp-server/rtsp-media.c:
6282 rtsp-media: Always use real payloader when creating streams
6283 A bin that contains the real payloader might be used as payloader. In this
6284 case we have to get the real payloader for the various properties it provides.
6285 Example use cases for this are bins that payload some media and then have
6286 additional elements that add metadata or RTP extension headers to the stream.
6287 https://bugzilla.gnome.org/show_bug.cgi?id=750800
6289 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
6291 * examples/test-netclock-client.c:
6292 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
6294 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
6296 * examples/test-netclock-client.c:
6297 * examples/test-netclock.c:
6298 test-netclock: Use new ntp-time-source property on rtpbin
6299 Select the clock time to be used as NTP time source. This allows proper
6300 synchronization between receivers, independent of sharing base times, and just
6301 requires them to use the same clock.
6303 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6305 * examples/test-netclock-client.c:
6306 * examples/test-netclock.c:
6307 test-netclock: Setting the same base time on sender and receiver is not necessary
6308 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
6310 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6312 * gst/rtsp-server/rtsp-stream.c:
6313 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
6314 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6316 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6318 * docs/libs/gst-rtsp-server.types:
6319 docs: add missing types
6320 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6322 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6324 * docs/libs/gst-rtsp-server-sections.txt:
6325 docs: add missing apis
6326 https://bugzilla.gnome.org/show_bug.cgi?id=750764
6328 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
6330 * examples/test-netclock-client.c:
6331 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
6333 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
6335 * docs/libs/gst-rtsp-server-sections.txt:
6336 * gst/rtsp-server/rtsp-auth.c:
6337 * gst/rtsp-server/rtsp-auth.h:
6338 GstRTSPAuth: Add client certificate authentication support
6339 https://bugzilla.gnome.org/show_bug.cgi?id=750471
6341 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
6343 * examples/test-netclock-client.c:
6344 test-netclock-client: Use new GstClock API to wait for clock synchronization
6346 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
6348 * examples/test-netclock-client.c:
6349 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
6350 A mainloop is needed to get glimagesink to display something on OSX, and
6351 the source-setup signal just makes things a little bit easier.
6353 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
6356 Automatic update of common submodule
6357 From d9a3353 to 6015d26
6359 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
6362 Automatic update of common submodule
6363 From d37af32 to d9a3353
6365 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
6368 Automatic update of common submodule
6369 From 21ba2e5 to d37af32
6371 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
6374 Automatic update of common submodule
6375 From c408583 to 21ba2e5
6377 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
6379 * docs/libs/Makefile.am:
6380 docs: remove variables that we define in the snippet from common
6381 This is syncing our Makefile.am with upstream gtkdoc.
6383 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6386 Automatic update of common submodule
6387 From 44a3517 to c408583
6389 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
6394 === release 1.5.1 ===
6396 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
6402 * gst-rtsp-server.doap:
6405 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
6407 * gst/rtsp-server/rtsp-client.c:
6408 rtsp-client: No flush during Teardown.
6409 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
6410 backlog is empty it can happen that just a part of a message will be
6411 sent and rest is in backlog queue. If then flush during teardown
6412 just a part of message will be sent.This can lead to client miss
6413 teardown response since it expect to get the last part of message.
6414 The flushing during teardown was introduced to fix a deadlock that now
6415 is fixed more generally in handle_request by temporary setting backlog
6417 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
6419 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
6421 * tests/check/Makefile.am:
6422 tests: Use AM_TESTS_ENVIRONMENT
6423 Needed by the new automake test runner and the
6424 current version of the common submodule.
6426 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
6428 * gst/rtsp-server/rtsp-media.h:
6429 * gst/rtsp-server/rtsp-stream.h:
6430 rtsp-server: Use single-include rtsp header to make sure we get all definitions
6432 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
6434 * gst/rtsp-server/rtsp-media.c:
6435 rtsp-media: Mark some more functions static
6437 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6439 * gst/rtsp-server/rtsp-media.c:
6440 rtsp-media: Only unblock the media in suspend() when actually changing the state
6441 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
6443 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6445 * examples/test-video-rtx.c:
6446 examples: Use AVPF profile for the RTX example
6448 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
6450 * gst/rtsp-server/rtsp-sdp.c:
6451 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
6453 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6455 * gst/rtsp-server/rtsp-stream.c:
6456 rtsp-stream: get valid clock-rate from last-sample
6457 clock-rate in last-sample's caps is integer, not unsigned.
6458 To get this value properly, variable needs to be type-casted to int.
6459 https://bugzilla.gnome.org/show_bug.cgi?id=747614
6461 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
6465 autogen.sh: only run autopoint if gettext requested in configure.ac
6466 Not just because there happens to be a po directory.
6467 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6469 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
6472 Revert "configure.ac: uncomment gettext version setup"
6473 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
6474 We don't need a gettext setup here and there's no po
6475 directory either, so no reason why autopoint would be
6476 run in the first place.
6477 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
6479 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
6481 * examples/test-multicast.c:
6482 * examples/test-multicast2.c:
6483 * examples/test-sdp.c:
6484 * examples/test-video-rtx.c:
6485 * examples/test-video.c:
6486 * tests/test-cleanup.c:
6487 * tests/test-reuse.c:
6488 Fix timeout function signatures across tests and examples
6490 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
6492 * tests/check/Makefile.am:
6493 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
6494 Make sure the test environment is set up.
6495 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6497 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
6500 configure: bump automake requirement to 1.14 and autoconf to 2.69
6501 This is only required for builds from git, people can still
6502 build tarballs if they only have older autotools.
6503 https://bugzilla.gnome.org//show_bug.cgi?id=747624
6505 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6508 configure.ac: uncomment gettext version setup
6509 Fixes autogen.sh. It would run autopoint, which would complain
6510 that it could not find the gettext version in configure.ac.
6511 https://bugzilla.gnome.org/show_bug.cgi?id=748058
6513 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6515 * examples/test-video-rtx.c:
6516 test-video-rtx: set exact payload type to PCMA payloader
6517 Setting wrong payload type causes failure to do retransmission through audio stream
6518 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6520 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
6522 * gst/rtsp-server/rtsp-media.c:
6523 * gst/rtsp-server/rtsp-stream.c:
6524 * gst/rtsp-server/rtsp-stream.h:
6525 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
6526 Because of duplicated g_signal_connect for request-aux-sender signal,
6527 wrong stream pointer is passed to the signal handler.
6528 Instead of passing each stream, pass stream array and get the relevant stream.
6529 https://bugzilla.gnome.org/show_bug.cgi?id=747839
6531 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
6535 Update autogen.sh to latest version from common
6536 Fixes build after aclocal_check etc. helpers have been removed.
6538 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
6541 Automatic update of common submodule
6542 From bc76a8b to c8fb372
6544 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6546 * gst/rtsp-server/rtsp-stream.c:
6547 rtsp-stream: Limit the queues to 1 buffer
6548 We only need them to be able to pre-roll, queueing up more data here
6549 is only going to harm latency and memory usage.
6551 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
6553 * gst/rtsp-server/rtsp-stream.c:
6554 rtsp-stream: Update comment and ASCII art to the latest code
6555 We have a queue in front of the udpsink too to prevent the pipeline from
6558 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6560 * gst/rtsp-server/rtsp-stream.c:
6561 rtsp-media: Properly return first rtptime
6562 Instead we where returning first GstBuffer timestamp. This would result
6563 in clock skew and unwanted behaviour in RTSP playback.
6564 https://bugzilla.gnome.org/show_bug.cgi?id=746479
6566 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
6568 * gst/rtsp-server/rtsp-stream.c:
6569 rtsp-stream: Don't leave buffer mapped
6570 If the seq is NULL, the RTP buffer was left mapped. We should always
6573 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
6578 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
6580 * gst/rtsp-server/rtsp-media-factory.c:
6581 * tests/check/gst/client.c:
6582 Fix double semicolons
6584 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
6586 * gst/rtsp-server/rtsp-stream.c:
6587 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
6588 This gives more accurate values than asking the payloader. There might be
6589 queueing happening between the payloader and the sink.
6590 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6592 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
6594 * gst/rtsp-server/rtsp-media.c:
6595 rtsp-media: Don't seek for PLAY if the position will not change
6596 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6598 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6600 * gst/rtsp-server/rtsp-media.c:
6601 rtsp-media: Don't include payload type in the caps for framesize
6602 When the sdp media attribute framesize are converted to caps
6603 the <payload> should not be included.
6604 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6605 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6607 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6609 * gst/rtsp-server/rtsp-sdp.c:
6610 rtsp-sdp: add payload type to the sdp framesize attribute
6611 The sdp framesize attribute is desribed in RFC6064. It is specified
6612 for payloading of H263 and has the following form
6613 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6614 should be added to the caps in a payloader and the <payload type> should
6615 be added by the rtsp-server.
6616 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6618 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6620 * examples/test-uri.c:
6621 examples: test-uri: fix tainted variable
6622 Insignificant but this keeps Coverity happy.
6625 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6627 * examples/.gitignore:
6628 * examples/Makefile.am:
6629 * examples/test-netclock-client.c:
6630 * examples/test-netclock.c:
6631 examples: Add a simple example of network synch for live streams.
6632 An example server and client that works for synchronising live streams
6633 only - as it can't support pause/play.
6635 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6637 * gst/rtsp-server/rtsp-media-factory.c:
6638 * gst/rtsp-server/rtsp-media-factory.h:
6639 rtsp-media-factory: Add functions to set/get the media gtype
6640 Allow specifying the GType of a GstRtspMedia subclass to create
6641 as a simpler way to get the factory to create a custom
6642 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6644 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6646 * gst/rtsp-server/rtsp-media.c:
6647 rtsp-media: fix double unlock in _get_buffer_size()
6648 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6649 because of double g_mutex_unlock () usage.
6650 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6652 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6654 * gst/rtsp-server/rtsp-session-pool.c:
6655 * gst/rtsp-server/rtsp-session.c:
6656 * gst/rtsp-server/rtsp-session.h:
6657 rtsp-session: Use monotonic time for RTSP session timeout
6658 Changed RTSP session timeout handling to monotonic time
6659 and deprecating the API for current system time.
6660 This fixes timeouts when the system time changes.
6661 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6663 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6665 * gst/rtsp-server/rtsp-client.c:
6666 * gst/rtsp-server/rtsp-media.c:
6667 rtsp-client: Only error out in PLAY if seeking actually failed
6668 If the media was just not seekable, we continue from whatever position we are
6669 and let the client decide if that is what is wanted or not.
6670 Only if the actual seek failed, we can't really recover and should error out.
6672 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6674 * gst/rtsp-server/rtsp-stream.c:
6675 rtsp-stream: Add necessary queues between tee and multiudpsink
6676 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6678 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6680 * gst/rtsp-server/rtsp-client.c:
6681 * gst/rtsp-server/rtsp-media.c:
6682 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6683 Instead error out properly the same way as if the SEEKING query already
6686 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6688 * gst/rtsp-server/rtsp-stream.h:
6689 rtsp-stream: minor code formatting fix
6691 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6693 * gst/rtsp-server/rtsp-media.c:
6694 rtsp-media: fix logic for collect_streams
6695 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6696 all streams it knows if it got any, and can check if the transport mode is OK.
6699 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6701 * gst/rtsp-server/rtsp-media.c:
6702 rtsp-media: Don't set the transport mode based on what elements we find
6703 Just print a warning if the one that was set before disagrees with what
6704 elements we found. It must already be set to something before as this
6705 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6706 and we would reject ANNOUNCE if the RECORD flag was not set.
6708 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6710 * tests/check/gst/rtspserver.c:
6711 tests: rtspserver: rename shadowed variable
6712 We have two different 'sink' variables here,
6713 rename one of them for clarity.
6715 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6717 * gst/rtsp-server/rtsp-client.c:
6718 rtsp-client: fix awkward if clause
6720 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6722 * examples/test-uri.c:
6723 examples: test-uri: improve uri argument handling and accept file names
6724 Print an error if the argument passed is not a URI and can't
6725 be converted into one, or no arguments have been provided.
6727 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6729 * examples/test-uri.c:
6730 examples: test-uri: don't remove mount point after 10 seconds
6731 It's very irritating when trying to test stuff repeatedly
6732 and serves no real purpose other than showing that it can
6735 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6737 * examples/.gitignore:
6738 examples: add new test-record to .gitignore
6740 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6742 * examples/test-record.c:
6743 * gst/rtsp-server/rtsp-client.c:
6744 * gst/rtsp-server/rtsp-media-factory.c:
6745 * gst/rtsp-server/rtsp-media-factory.h:
6746 * gst/rtsp-server/rtsp-media.c:
6747 * gst/rtsp-server/rtsp-media.h:
6748 * tests/check/gst/rtspserver.c:
6749 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6751 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6753 * examples/test-record.c:
6754 test-record: Set latency for playback-style example to 2s instead of 200ms
6756 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6758 * tests/check/gst/rtspserver.c:
6759 tests: add some unit tests for ANNOUNCE and RECORD
6760 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6762 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6764 * gst/rtsp-server/rtsp-client.c:
6765 rtsp-client: fix a couple of leaks in handle_announce
6767 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6769 * gst/rtsp-server/rtsp-media-factory.c:
6770 * gst/rtsp-server/rtsp-media-factory.h:
6771 * gst/rtsp-server/rtsp-media.c:
6772 * gst/rtsp-server/rtsp-media.h:
6773 rtsp-media: Expose latency setting for setting the rtpbin latency
6775 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6777 * examples/test-record.c:
6778 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6780 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6782 * gst/rtsp-server/rtsp-stream.c:
6783 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6785 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6787 * examples/Makefile.am:
6788 * examples/test-record.c:
6789 * gst/rtsp-server/rtsp-client.c:
6790 * gst/rtsp-server/rtsp-client.h:
6791 * gst/rtsp-server/rtsp-media-factory.c:
6792 * gst/rtsp-server/rtsp-media-factory.h:
6793 * gst/rtsp-server/rtsp-media.c:
6794 * gst/rtsp-server/rtsp-media.h:
6795 * gst/rtsp-server/rtsp-session-media.c:
6796 * gst/rtsp-server/rtsp-stream.c:
6797 * gst/rtsp-server/rtsp-stream.h:
6798 Add initial support for RECORD
6799 We currently only support media that is RECORD or PLAY only, not both at once.
6800 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6802 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6804 * gst/rtsp-server/rtsp-stream.c:
6805 rtsp-stream: RTCP and RTP transport cache cookies seperated
6806 RTCP packets were not sent because the same tr_cache_cookie was used for
6807 both RTP and RTCP. So only one of the tr_cache lists were populated
6808 depending on which one was sent first. If the tr_cache list is not
6809 populated then no packets can be sent. Most often this happened to be
6810 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6811 resulted in both the tr_cache_lists to be populated regardless of which
6813 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6815 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6817 * gst/rtsp-server/rtsp-stream.c:
6818 rtsp-stream: fix false compiler warning
6819 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6821 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6823 * gst/rtsp-server/rtsp-client.c:
6824 rtsp-client: log interleaved data received
6826 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6828 * gst/rtsp-server/rtsp-client.c:
6829 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6831 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6833 * gst/rtsp-server/rtsp-client.c:
6834 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6836 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6838 * gst/rtsp-server/rtsp-client.c:
6839 rtsp-client: Use a random session ID in the SDP
6840 RFC4566 Section 5.2 says that it should make the username, session id,
6841 nettype, addrtype and unicast address tuple globally unique. Always using
6842 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6843 Instead let's create a 64 bit random number, which at least brings us
6844 closer to the goal of global uniqueness.
6845 https://tools.ietf.org/html/rfc4566#section-5.2
6847 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6849 * examples/test-launch.c:
6850 * examples/test-mp4.c:
6851 * examples/test-ogg.c:
6852 * examples/test-uri.c:
6853 examples: Don't call gst_init() and gst_get_option_group()
6854 The latter calls the former at the appropriate time.
6856 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6858 * gst/rtsp-server/rtsp-client.c:
6859 rtsp-client: Drop trailing \0 of RTSP DATA messages
6860 We add a trailing \0 in GstRTSPConnection to make parsing of
6861 string message bodies easier (e.g. the SDP from DESCRIBE) but
6862 for actual data this means we have to drop it or otherwise
6863 create invalid data.
6865 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6867 * gst/rtsp-server/rtsp-stream.c:
6868 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6869 Fixes crash when two threads access handle_new_sample() at the same
6870 time, one for RTP, one for RTCP.
6871 Otherwise, when iterating over the transports cache, it might be modified by
6872 another thread at the same time if the transports cookie has changed.
6873 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6875 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6877 * gst/rtsp-server/rtsp-stream.c:
6878 rtsp-stream: Set format=TIME on our app sources for TCP
6880 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6882 * gst/rtsp-server/rtsp-session-pool.c:
6883 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6884 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6885 RFC 2326 states that session IDs may consist of alphanumeric as well as
6886 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6887 Previously the session ID was URI-escaped, this meant that any character
6888 which was not alphanumeric or any of the characters +-._~ would be
6889 percent encoded. While the RFC (surprisingly) mentions that linear white
6890 space in session IDs should be URI-escaped, it does not say anything
6891 about other characters. Moreover no white space is allowed in the
6892 session ID. Finally the percent character which is the result of
6893 URI-escaping is not allowed in a session ID.
6894 So there is no reason to do any URI-escaping, and now it is removed.
6895 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6897 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6900 Automatic update of common submodule
6901 From f2c6b95 to bc76a8b
6903 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6906 Fix 'make check' from top-level directory
6908 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6910 * examples/test-launch.c:
6911 * examples/test-mp4.c:
6912 * examples/test-ogg.c:
6913 * examples/test-uri.c:
6914 examples: Add command-line parsing and take a 'port' argument
6915 This allows users to run multiple servers on different ports for testing.
6916 Only done for examples that actually take arguments and hence are capable of
6917 outputting different streams for each instance on each port.
6918 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6920 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-client.h:
6924 rtsp-client: Add a send_message default signal handler
6925 This allows subclasses to easily hook into the response sending
6926 mechanism without doing everything from a signal, which seems
6927 awkward from subclasses.
6929 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6932 Automatic update of common submodule
6933 From ef1ffdc to f2c6b95
6935 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6939 configure: add --disable-examples switch
6940 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6942 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6944 * examples/.gitignore:
6945 * examples/Makefile.am:
6946 * examples/test-video-rtx.c:
6947 examples: add a retransmisison example implementing RFC4588
6948 Currently only SSRC-multiplexed rtx streams are supported
6950 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6952 * gst/rtsp-server/rtsp-stream.c:
6953 rtsp-stream: Fix some minor memory leaks
6955 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6957 * gst/rtsp-server/rtsp-media.c:
6958 rtsp-media: Some minor cleanup
6960 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6962 * gst/rtsp-server/rtsp-stream.c:
6963 rtsp-stream: Fix compiler warnings
6964 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6965 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6967 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6968 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6971 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6973 * docs/libs/gst-rtsp-server-sections.txt:
6974 * gst/rtsp-server/rtsp-media-factory.c:
6975 * gst/rtsp-server/rtsp-media-factory.h:
6976 * gst/rtsp-server/rtsp-media.c:
6977 * gst/rtsp-server/rtsp-media.h:
6978 * gst/rtsp-server/rtsp-sdp.c:
6979 * gst/rtsp-server/rtsp-stream.c:
6980 * gst/rtsp-server/rtsp-stream.h:
6981 media: implement ssrc-multiplexed retransmission support
6982 based off RFC 4588 and the server-rtpaux example in -good
6984 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6986 * gst/rtsp-server/rtsp-client.c:
6987 * gst/rtsp-server/rtsp-stream-transport.c:
6988 * gst/rtsp-server/rtsp-stream.c:
6989 rtsp: Ref transports in hash table.
6990 Also ref streams for transports.
6991 This solves a crash when reciving a rtcp after teardown but before
6993 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6995 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6998 Automatic update of common submodule
6999 From 7bb2bce to ef1ffdc
7001 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
7003 * gst/rtsp-server/rtsp-client.c:
7004 client: refactor cleanup of cached media
7006 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
7008 * tests/check/gst/client.c:
7010 The session leak is now fixed, lets remove those FIXME comments.
7012 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
7014 * tests/check/gst/rtspserver.c:
7015 tests: Test to setup two sessions on one connection
7016 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7018 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
7020 * tests/check/gst/rtspserver.c:
7021 tests: Test setup with tcp transport
7022 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7024 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
7026 * gst/rtsp-server/rtsp-client.c:
7027 client: Configure transport after creating session media
7028 The default implementation of configure_client_transport() in
7029 rtsp-client uses the session media when it chooses channels for
7030 interleaved traffic.
7031 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7033 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
7035 * gst/rtsp-server/rtsp-client.c:
7036 * gst/rtsp-server/rtsp-session-media.c:
7037 client: Stop caching media in client when doing setup
7038 If the media has been managed by a session media, it should not be
7039 cached in the client any longer. The GstRTSPSessionMedia object is now
7040 responsible for unpreparing the GstRTSPMedia object using
7041 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
7043 https://bugzilla.gnome.org/show_bug.cgi?id=739112
7045 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7047 * gst/rtsp-server/rtsp-stream.c:
7048 rtsp-stream: unref srtp decoder when leaving bin
7049 https://bugzilla.gnome.org/show_bug.cgi?id=739481
7051 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7053 * gst/rtsp-server/rtsp-client.c:
7054 rtsp-client: mikey memory leaks
7055 https://bugzilla.gnome.org/show_bug.cgi?id=739383
7057 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
7060 Automatic update of common submodule
7061 From 84d06cd to 7bb2bce
7063 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
7066 Parallelise 'make check-valgrind'
7068 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
7071 Automatic update of common submodule
7072 From a8c8939 to 84d06cd
7074 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
7077 Automatic update of common submodule
7078 From 36388a1 to a8c8939
7080 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7082 * gst/rtsp-server/rtsp-media.c:
7083 rtsp-media: deactivate media when shutting down from paused
7084 This was only done when going directly from playing.
7085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
7087 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7089 * gst/rtsp-server/rtsp-client.c:
7090 * gst/rtsp-server/rtsp-context.h:
7091 rtsp-client: add stream transport to context
7092 We add the stream transport to the context so we can get the configured
7093 client stream transport in the setup request signal.
7094 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
7096 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7098 * gst/rtsp-server/rtsp-stream.c:
7099 stream: release lock even not all transports have been removed
7100 We don't want to keep the lock even we return FALSE because not all the
7101 transports have been removed. This could lead into a deadlock.
7102 https://bugzilla.gnome.org/show_bug.cgi?id=737797
7104 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
7106 * gst/rtsp-server/rtsp-sdp.c:
7107 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
7108 These were renamed in GstRTPBasePayload in 1.0
7110 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7112 * gst/rtsp-server/rtsp-client.c:
7113 client: set session media to NULL without the lock
7114 We need to set session medias to NULL without the client lock otherwise
7115 we can end up in a deadlock if another thread is waiting for the lock
7116 and media unprepare is also waiting for that thread to end.
7117 https://bugzilla.gnome.org/show_bug.cgi?id=737690
7119 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
7121 * gst/rtsp-server/rtsp-media.c:
7122 rtsp-media: Set state to UNPREPARING in all cases
7124 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
7126 * gst/rtsp-server/rtsp-media.c:
7127 media: set state to unpreparing when unprepare is initiated
7128 https://bugzilla.gnome.org/show_bug.cgi?id=737675
7130 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
7132 * gst/rtsp-server/rtsp-client.c:
7133 rtsp-client: Remove backlog limit while processings requests
7134 If the backlog limit is kept two cases of deadlocks may be
7135 encountered when streaming over TCP. Without the backlog
7136 limit this deadlocks can not happen, at the expence of
7138 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
7140 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
7142 * gst/rtsp-server/rtsp-client.c:
7143 rtsp-client: do not free main context before rtsp watch
7144 https://bugzilla.gnome.org/show_bug.cgi?id=737110
7146 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
7148 * tests/check/gst/rtspserver.c:
7149 tests: Extend unit test timeout to accomodate for valgrind
7150 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7152 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
7154 * gst/rtsp-server/rtsp-client.c:
7155 * gst/rtsp-server/rtsp-session.c:
7156 * gst/rtsp-server/rtsp-stream-transport.c:
7157 rtsp-*: Treat sending packets to clients as keepalive
7158 As long as gst-rtsp-server can successfully send RTP/RTCP data to
7159 clients then the client must be reading. This change makes the server
7160 timeout the connection if the client stops reading.
7161 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7163 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
7165 * gst/rtsp-server/rtsp-client.c:
7166 rtsp-client: Allow backlog to grow while expiring session
7167 Allow the send backlog in the RTSP watch to grow to unlimited size while
7168 attempting to bring the media pipeline to NULL due to a session
7169 expiring. Without this change the appsink element cannot change state
7170 because it is blocked while rendering data in the new_sample callback.
7171 This callback will block until it has successfully put the data into the
7172 send backlog. There is a chance that the send backlog is full at this
7173 point which means that the callback may block for a long time, possibly
7174 forever. Therefore the media pipeline may also be prevented from
7175 changing state for a long time.
7176 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
7178 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
7180 * gst/rtsp-server/rtsp-client.c:
7181 rtsp-client: Make old compilers happy
7182 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
7183 Just in case that guint8 doesn't fit in a pointer. Just in case ...
7185 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
7187 * gst/rtsp-server/rtsp-client.c:
7188 client: raise the backlog limits before pausing
7189 We need to raise the backlog limits before pausing the pipeline or else
7190 the appsink might be blocking in the render method in wait_backlog() and
7191 we would deadlock waiting for paused.
7192 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
7194 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
7196 * gst/rtsp-server/rtsp-client.c:
7197 client: make define for the WATCH_BACKLOG
7198 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
7200 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
7202 * gst/rtsp-server/rtsp-client.c:
7203 client: simplify session transport handling
7204 link/unlink of the transport in a session was done to keep track of all
7205 TCP transports and to send RTP/RTCP data to the streams. We can simplify
7206 that by putting all the TCP transports in a hashtable indexed with the
7208 We also don't need to link/unlink the transports when we pause/resume
7209 the streams. The same effect is already achieved when we pause/play the
7210 media. Indeed, when we pause the media, the transport is removed from
7211 the media and the callbacks will not be called anymore.
7212 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
7214 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
7216 * gst/rtsp-server/rtsp-stream-transport.c:
7217 * gst/rtsp-server/rtsp-stream-transport.h:
7218 stream-transport: make method to handle received data
7219 Make a method to handle the data received on a channel. It sends the
7220 data to the stream of the transport on the RTP or RTCP pads based on
7223 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
7225 * examples/test-mp4.c:
7226 test: add example of dumping RTCP reports
7228 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
7230 * gst/rtsp-server/rtsp-media.c:
7231 * gst/rtsp-server/rtsp-stream.c:
7232 * gst/rtsp-server/rtsp-stream.h:
7233 rtsp-media: Make sure that sequence numbers are monotonic after pause
7234 The sequence number is not monotonic for RTP packets after pause. The
7235 reason is basepayloader generates a randon sequence number when the
7236 pipeline goes from ready to pause. With this fix generation of sequence
7237 number will be monotonic when going from pause to play request.
7238 https://bugzilla.gnome.org/show_bug.cgi?id=736017
7240 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
7242 * gst/rtsp-server/rtsp-client.c:
7243 rtsp-client: Protect saved clients watch with a mutex
7244 Fixes a crash when close() is called while merging clients
7245 in handle_tunnel(). In that case close() would destroy the
7246 watch while it is still being used in handle_tunnel().
7247 https://bugzilla.gnome.org/show_bug.cgi?id=735570
7249 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
7251 * gst/rtsp-server/rtsp-stream.c:
7252 rtsp-stream: Remove the multicast group udp sources when removing from the bin
7254 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
7256 * gst/rtsp-server/rtsp-media.c:
7257 * gst/rtsp-server/rtsp-stream.c:
7258 * gst/rtsp-server/rtsp-stream.h:
7259 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
7260 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
7261 seeking and will always continue counting the time. This leads to
7262 the NPT after a backwards seek to be something completely different
7263 to the actual seek position.
7264 https://bugzilla.gnome.org/show_bug.cgi?id=732644
7266 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
7268 * examples/test-appsrc.c:
7269 examples: fix another reference leak
7270 gst_rtsp_media_get_element() returns a new ref.
7272 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
7274 * examples/test-appsrc.c:
7275 examples: unref element after usage
7276 gst_bin_get_by_name_recurse_up() returns an element
7277 reference that must be unreffed after usage.
7278 https://bugzilla.gnome.org/show_bug.cgi?id=734546
7280 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
7282 * gst/rtsp-server/rtsp-media.c:
7283 signals: Fix copy-pasto in target-state signal offset
7285 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
7289 Makefile: Add usage of build-checks step
7290 Allows building checks without running them
7292 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
7294 * gst/rtsp-server/rtsp-stream.c:
7295 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
7296 When a UDP multicast transport is used it is expected that the server listens
7297 for RTP and RTCP packets on the multicast group with the corresponding port.
7298 Without this we will never get RTCP packets from clients in multicast mode.
7299 https://bugzilla.gnome.org/show_bug.cgi?id=732238
7301 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
7306 === release 1.4.0 ===
7308 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7314 * gst-rtsp-server.doap:
7317 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
7319 * gst/rtsp-server/rtsp-media.h:
7320 media: correct misspelled words in description
7321 https://bugzilla.gnome.org/show_bug.cgi?id=733244
7323 === release 1.3.91 ===
7325 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
7331 * gst-rtsp-server.doap:
7334 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
7336 * docs/libs/gst-rtsp-server-sections.txt:
7339 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
7341 * gst/rtsp-server/rtsp-server.c:
7342 server: implement client REMOVE filter
7344 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
7346 * gst/rtsp-server/rtsp-client.c:
7347 * gst/rtsp-server/rtsp-client.h:
7348 client: expose _close() method
7349 Expose a previously internal close method to close the client
7352 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
7354 * gst/rtsp-server/rtsp-session-pool.c:
7355 session-pool: signal session-removed outside of the lock
7356 Release the lock before emiting the session-removed signal.
7358 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
7360 * gst/rtsp-server/rtsp-client.c:
7361 * gst/rtsp-server/rtsp-server.c:
7362 * gst/rtsp-server/rtsp-session-pool.c:
7363 * gst/rtsp-server/rtsp-session.c:
7364 * gst/rtsp-server/rtsp-stream.c:
7365 filter: Release lock in filter functions
7366 Release the object lock before calling the filter functions. We need to
7367 keep a cookie to detect when the list changed during the filter
7368 callback. We also keep a hashtable to make sure we only call the filter
7369 function once for each object in case of concurrent modification.
7370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
7372 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
7374 * gst/rtsp-server/rtsp-client.c:
7375 client: check if watch is set in handle_teardown()
7376 The unit tests run without a watch
7378 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
7380 * tests/check/gst/client.c:
7381 client tests: send teardown to cleanup session
7383 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
7385 * tests/check/gst/rtspserver.c:
7386 server tests: send teardown to cleanup session
7388 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7390 * gst/rtsp-server/rtsp-client.c:
7391 client: keep ref to client for the session removed handler
7392 This extra ref will be dropped when all client sessions have been
7393 removed. A session is removed when a client sends teardown, closes its
7394 endpoint of the TCP connection or the sessions expires.
7395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7397 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
7399 * gst/rtsp-server/rtsp-client.c:
7400 * gst/rtsp-server/rtsp-session.c:
7401 * tests/check/gst/client.c:
7402 client: manage media in session as a last step
7403 Once we manage a media in a session, we can't unmanage it anymore
7404 without destroying it. Therefore, first check everything before we
7405 manage the media, otherwise if something is wrong we have no way to
7407 If we created a new session and something went wrong, remove the session
7408 again. Fixes a leak in the unit test.
7410 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
7412 * examples/test-mp4.c:
7413 * examples/test-ogg.c:
7414 examples: print 'stream ready at url' for mp4 and ogg example
7416 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
7418 * gst/rtsp-server/rtsp-client.c:
7419 * gst/rtsp-server/rtsp-sdp.c:
7420 rtsp: fix for MIKEY api change
7422 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
7424 * gst/rtsp-server/rtsp-client.c:
7425 client: free watch context only once
7426 The watch context is freed when the source is destroyed. Avoids
7427 a CRITICAL when we try to unref the context twice.
7429 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
7431 * gst/rtsp-server/rtsp-client.c:
7434 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
7436 * gst/rtsp-server/rtsp-client.c:
7437 client: protect sessions with lock
7438 Protect the list of sessions with the lock.
7439 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
7441 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
7443 * gst/rtsp-server/rtsp-client.c:
7444 Client: keep a ref to the session
7445 Don't just keep a weak ref to the session objects but use a hard ref. We
7446 will be notified when a session is removed from the pool (expired) with
7447 the new session-removed signal.
7448 Don't automatically close the RTSP connection when all the sessions of
7449 a client are removed, a client can continue to operate and it can create
7450 a new session if it wants. If you want to remove the client from the
7451 server, you have to use gst_rtsp_server_client_filter() now.
7452 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
7453 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
7455 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
7457 * gst/rtsp-server/rtsp-session-pool.c:
7458 * gst/rtsp-server/rtsp-session-pool.h:
7459 session-pool: add session-removed signal
7460 Add a signal to be notified when a session is removed from the pool.
7462 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
7464 * gst/rtsp-server/Makefile.am:
7465 * gst/rtsp-server/rtsp-server.h:
7466 Make rtsp-server.h a single-include header, use it for G-I
7467 https://bugzilla.gnome.org/show_bug.cgi?id=732411
7469 === release 1.3.90 ===
7471 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
7477 * gst-rtsp-server.doap:
7480 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
7482 * gst/rtsp-server/rtsp-stream.c:
7483 stream: crypto can be NULL
7485 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
7487 * gst/rtsp-server/rtsp-client.c:
7488 * gst/rtsp-server/rtsp-media.c:
7489 * gst/rtsp-server/rtsp-mount-points.c:
7490 introspection: add missing allow-none annotations
7491 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7493 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
7495 * gst/rtsp-server/rtsp-address-pool.c:
7496 * gst/rtsp-server/rtsp-media.c:
7497 * gst/rtsp-server/rtsp-session-media.c:
7498 * gst/rtsp-server/rtsp-session-pool.c:
7499 * gst/rtsp-server/rtsp-stream-transport.c:
7500 * gst/rtsp-server/rtsp-stream.c:
7501 * gst/rtsp-server/rtsp-token.c:
7502 introspection: add (nullable) annotations to return values
7503 https://bugzilla.gnome.org/show_bug.cgi?id=730952
7505 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
7507 * gst/rtsp-server/rtsp-client.c:
7508 * gst/rtsp-server/rtsp-stream.c:
7509 gi: improve annotations
7510 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
7512 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
7514 * gst/rtsp-server/rtsp-client.c:
7515 * gst/rtsp-server/rtsp-media-factory.c:
7516 * gst/rtsp-server/rtsp-media.c:
7517 * gst/rtsp-server/rtsp-server.c:
7518 signals: use generic marshal function
7519 Use the generic C marshal function.
7520 Use more explicit type instead of G_TYPE_POINTER
7522 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
7524 * gst/rtsp-server/rtsp-context.h:
7525 context: add type macro
7527 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
7529 * gst/rtsp-server/rtsp-client.c:
7530 * gst/rtsp-server/rtsp-sdp.c:
7531 * gst/rtsp-server/rtsp-sdp.h:
7532 sdp: hide key length defines
7533 They don't have a namespace.
7535 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
7540 === release 1.3.3 ===
7542 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
7548 * gst-rtsp-server.doap:
7551 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7553 * gst/rtsp-server/rtsp-client.c:
7554 * gst/rtsp-server/rtsp-sdp.c:
7555 * gst/rtsp-server/rtsp-sdp.h:
7556 mikey: add different key length parameters
7557 Add encryption and authentication key length parameters to MIKEY. For
7558 the encoders, the key lengths are obtained from the cipher and auth
7559 algorithms set in the caps. For the decoders, they are obtained while
7560 parsing the key management from the client.
7561 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
7563 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
7565 * tests/check/gst/stream.c:
7566 stream tests: Make sure we get right multicast address from stream
7567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
7569 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7571 * gst/rtsp-server/rtsp-client.c:
7572 client: ref the context until rtsp watch is alive
7573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
7575 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7577 * gst/rtsp-server/rtsp-client.c:
7578 client: Destroy the rtsp watch after connection close
7580 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
7582 * gst/rtsp-server/rtsp-media.c:
7583 media: fix confusing comment
7585 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
7587 * gst/rtsp-server/rtsp-session.c:
7588 rtsp-session: Timeout in header.
7589 Adding the possbilty to always have timout in header.
7590 This is configurabe with setting "timeout-always-visible".
7591 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
7593 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7598 === release 1.3.2 ===
7600 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7607 * gst-rtsp-server.doap:
7610 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7613 Automatic update of common submodule
7614 From 211fa5f to 1f5d3c3
7616 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7618 * gst/rtsp-server/rtsp-client.c:
7619 client: store TCP ports in transport
7620 Store the TCP ports in the transport when we are doing RTSP over TCP.
7621 This way, we can easily get to the ports from the transport.
7622 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7624 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7626 * gst/rtsp-server/rtsp-stream.c:
7627 stream: add signals for new RTP/RTCP encoders
7628 New signals to allow the user to configure the dynamically created
7630 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7632 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7634 * gst/rtsp-server/rtsp-media.c:
7635 * gst/rtsp-server/rtsp-media.h:
7636 media: Make suspend()/unsuspend() virtual
7637 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7639 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7641 * gst/rtsp-server/rtsp-client.c:
7642 client: fix send-message signal marshaller
7643 Use generic marshalling for the send-message signal. It has
7644 two POINTER arguments, not just one.
7645 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7647 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7649 * tests/check/gst/media.c:
7650 tests: add and remove pads only once
7651 In this test we simulate a dynamic pad by watching the caps event.
7652 Because of renegotiation in the base payloader now, this caps is sent
7653 multiple times but we can only deal with 1 invocation, use a variable to
7654 only 'add and remove' the pad once.
7656 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7658 * tests/check/gst/rtspserver.c:
7659 tests: add unit test for correct handling of Require headers
7660 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7662 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7664 * gst/rtsp-server/rtsp-client.c:
7665 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7666 Servers must handle Require headers and must report a failure
7667 if they don't handle any of the Required options, see RFC 2326,
7668 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7669 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7671 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7676 === release 1.3.1 ===
7678 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7684 * gst-rtsp-server.doap:
7687 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7690 Automatic update of common submodule
7691 From bcb1518 to 211fa5f
7693 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7698 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7700 * tests/check/gst/sessionmedia.c:
7701 tests: fix memory leak in sessionmedia unit test
7703 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7705 * gst/rtsp-server/rtsp-client.c:
7706 client: emit a signal before sending a message
7707 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7709 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7711 * gst/rtsp-server/rtsp-client.c:
7712 client: pass context to send_message
7713 Pass the current context to send_message, we will need it later.
7715 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7717 * gst/rtsp-server/rtsp-client.c:
7718 client: fix typo in comment
7720 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7722 * gst/rtsp-server/rtsp-media.c:
7723 media: Do not stop thread twice if default_prepare() fails
7725 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7727 * gst/rtsp-server/rtsp-client.c:
7728 client: set the watch to flushing before going to NULL
7729 First set the watch to flushing so that we unblock any current and
7730 future attempt to send data on the watch, Then set the pipeline to
7732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7734 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7736 * gst/rtsp-server/rtsp-session-pool.c:
7737 * tests/check/gst/sessionpool.c:
7738 rtsp-session-pool: Fixes annotation
7739 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7740 in the sessionpool test.
7741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7743 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7745 * gst/rtsp-server/rtsp-media.c:
7746 * gst/rtsp-server/rtsp-media.h:
7747 media: make media_prepare virtual
7748 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7750 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7752 * gst/rtsp-server/rtsp-media.c:
7753 * tests/check/gst/media.c:
7754 media: stop the thread in more error cases
7756 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7758 * gst/rtsp-server/rtsp-media.c:
7759 * tests/check/gst/media.c:
7760 media: allow NULL as the thread
7761 Use the default context whan passing a NULL thread.
7763 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7765 * gst/rtsp-server/rtsp-client.c:
7766 rtsp-client: indent cleanup
7767 Coverity was moaning about unreachable code, and I think it was just
7768 confused by { being before the label. We'll see if it pops up again.
7771 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7773 * gst/rtsp-server/rtsp-client.c:
7774 * gst/rtsp-server/rtsp-media.c:
7775 client: Add drop-backlog property
7776 When we have too many messages queued for a client (currently hardcoded
7777 to 100) we overflow and drop the messages. Add a drop-backlog property
7778 to control this behaviour. Setting this property to FALSE will retry
7779 to send the messages to the client by waiting for more room in the
7781 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7783 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7785 * gst/rtsp-server/rtsp-client.c:
7786 client: support for POST before GET when setting up a tunnel
7788 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7790 * gst/rtsp-server/rtsp-client.c:
7791 client: remove watch of the second client after http tunnel setup
7792 The second client will be freed after the HTTP tunnel has been set up.
7793 Make sure it's RTSP watch is never dispatched again.
7794 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7796 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7798 * gst/rtsp-server/rtsp-media.c:
7799 * tests/check/gst/media.c:
7800 media: Make media_prepare() fail if port allocation fails
7801 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7803 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7805 * tests/check/gst/media.c:
7806 media test: cleanup the thread pool in tests
7808 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7810 * gst/rtsp-server/rtsp-media.c:
7811 * tests/check/gst/media.c:
7812 rtsp-media: Unblock blocked streams in unprepare
7813 The streams will be blocked when a live media is prepared.
7814 The streams should be unblocked in gst_rtsp_media_unprepare.
7815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7817 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7819 * gst/rtsp-server/rtsp-media.c:
7820 media: release the state lock when going to NULL
7821 Set our state to UNPREPARING and release the state-lock before
7822 setting the pipeline to the NULL state. This way, any pad-added
7823 callback will be able to take the state-lock and check that we are now
7824 unpreparing instead of deadlocking.
7825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7827 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7829 * gst/rtsp-server/rtsp-media.c:
7830 media: protect status with lock
7831 Make sure we only update the status with the lock.
7833 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7835 * gst/rtsp-server/rtsp-client.c:
7836 * gst/rtsp-server/rtsp-sdp.c:
7837 rtsp: update for MIKEY API changes
7839 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7841 * gst/rtsp-server/rtsp-client.c:
7842 client: parse the mikey response from the client
7843 Parse the mikey response from the client and update the policy for
7846 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7848 * gst/rtsp-server/rtsp-stream.c:
7849 * gst/rtsp-server/rtsp-stream.h:
7850 stream: add method to set crypto info
7851 Make a method to configure the crypto information of a stream.
7852 Set udpsrc in READY instead of PAUSED so that we can configure caps
7855 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7857 * gst/rtsp-server/rtsp-client.c:
7858 client: cleanup error paths
7860 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7862 * gst/rtsp-server/rtsp-media.c:
7865 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7867 * examples/test-video.c:
7868 test: enable SRTP only on RTSPS
7869 We only want to enable SRTP when doing rtsp over TLS so that we can
7870 exchange the keys in a secure way.
7872 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7874 * examples/test-video.c:
7875 test: print an error on failure
7877 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7880 * examples/test-video.c:
7881 * gst/rtsp-server/rtsp-sdp.c:
7882 * gst/rtsp-server/rtsp-stream.c:
7883 * tests/check/Makefile.am:
7884 stream: add SRTP support
7885 Install srtp encoder and decoder elements in rtpbin
7888 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7890 * tests/check/Makefile.am:
7891 * tests/check/gst/sessionpool.c:
7892 tests: Add unit tests for sessionpool
7893 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7895 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7897 * tests/check/gst/threadpool.c:
7898 tests: Improve code coverage of rtsp-threadpool tests
7899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7901 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7903 * tests/check/gst/sessionmedia.c:
7904 tests: Improve code coverage for rtsp-session-media
7905 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7907 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7909 gobject-introspection: Add annotations to support language bindings
7910 In addition a few cosmetic changes:
7911 * Adjust the order of arguments
7912 * Fix typo: occured -> occurred
7913 * Fix indentation after Return:-clauses
7914 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7916 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7918 * gst/rtsp-server/rtsp-stream.c:
7919 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7920 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7922 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7924 * gst/rtsp-server/rtsp-stream.c:
7925 stream: take caps after the session manager
7926 Take the caps for the SDP after they leave the rtpbin so that we can
7927 also get the properties added by rtpbin elements.
7929 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7931 * gst/rtsp-server/rtsp-stream.c:
7932 stream: release lock while pushing out packets
7933 Keep a cache of the transports and use this to iterate the transport
7934 while pushing packets. This allows us to release the lock early.
7935 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7937 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7939 * gst/rtsp-server/rtsp-client.c:
7940 * gst/rtsp-server/rtsp-client.h:
7941 rtsp-client: vmethod for modifying tunnel GET response
7942 Add a vmethod tunnel_http_response where the response to the HTTP GET
7943 for tunneled connections can be modified.
7944 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7946 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7948 * gst/rtsp-server/rtsp-sdp.c:
7949 sdp: make 1 media line per profile
7950 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7951 line in the SDP for each profile. The client is then supposed to pick
7952 one of the profiles in the SETUP request. Because the m= lines have the
7953 same pt, the client also knows that only 1 option is possible.
7955 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7957 * gst/rtsp-server/rtsp-media-factory.c:
7958 * gst/rtsp-server/rtsp-media-factory.h:
7959 * gst/rtsp-server/rtsp-media.c:
7960 factory: add profile property and pass to media and streams
7962 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7964 * examples/test-multicast.c:
7965 * gst/rtsp-server/rtsp-sdp.c:
7966 sdp: pass multicast connection for multicast-only stream
7967 Pass the multicast address of the stream in the connection info in the
7968 SDP so that clients try a multicast connection first.
7969 Only allow multicast connections in the test-multicast example. Also
7970 increase the TTL a little.
7972 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7975 .gitignore: Ignore gcov intermediate files
7976 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7978 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7980 * gst/rtsp-server/rtsp-stream.c:
7981 stream: release some locks in error cases
7983 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7985 docs: Enable and fix gtk-doc warnings
7986 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7987 * addresspool/mediafactory: Add missing annotation colon
7988 * stream: Annotate return value
7989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7991 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7994 Automatic update of common submodule
7995 From fe1672e to bcb1518
7997 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
8000 Automatic update of common submodule
8001 From 1a07da9 to fe1672e
8003 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
8005 * examples/Makefile.am:
8006 examples: use LDADD for libs instead of LDFLAGS
8008 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
8011 configure: make sure releases are in .doap file
8013 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
8015 * examples/test-cgroups.c:
8016 examples: test-cgroups: don't put code with side effects into g_assert()
8017 The g_assert() might get compiled out with the right
8018 compiler/preprocessor flags.
8020 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
8022 * examples/.gitignore:
8023 examples: add cgroup test binary to .gitignore
8025 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
8027 * examples/test-cgroups.c:
8028 examples: fix cgroup test build
8029 Fixes build failure caused by compiler warning:
8030 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
8032 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
8035 .gitignore: ignore temp files created in the course of 'make check'
8037 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
8039 * gst/rtsp-server/rtsp-media.c:
8040 rtsp-media: don't loose frames handling new PLAY request
8041 If client supplied a range check if the range specifies the start point.
8042 If not, then do an accurate seek to the current position. If a start
8043 point was specified do do a key unit seek to make sure the streaming
8044 starts with decodeable frames.
8045 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
8047 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
8049 * gst/rtsp-server/rtsp-media.c:
8050 Revert "media: only flush when setting a new start position"
8051 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
8052 We need to do the flush in all cases, demuxer block currently for
8055 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
8057 * gst/rtsp-server/rtsp-media.c:
8058 media: only flush when setting a new start position
8059 Only flush the pipeline when we change the start position with
8061 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
8063 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
8065 * gst/rtsp-server/rtsp-stream.c:
8066 stream: set ttl-mc before adding the socket
8067 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
8068 never be set on socket.
8069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
8071 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8073 * gst/rtsp-server/rtsp-media.c:
8074 media: stop thread if media is already prepared
8075 in gst_rtsp_media_prepare() the thread is not used if media is already
8076 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
8078 https://bugzilla.gnome.org/show_bug.cgi?id=724182
8080 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
8083 build: Ship gst-rtsp-server.doap file
8085 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
8087 * tests/check/gst/rtspserver.c:
8088 tests: Fix another compiler warning with gcc
8090 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
8092 * gst/rtsp-server/rtsp-client.c:
8093 * gst/rtsp-server/rtsp-mount-points.c:
8094 * gst/rtsp-server/rtsp-stream.c:
8095 * tests/check/gst/client.c:
8096 rtsp-server: Fix lots of compiler warnings with clang
8098 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
8101 * gst-rtsp-server.doap:
8102 * tests/Makefile.am:
8103 configure: Synchronise with the configure scripts of the other modules
8105 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
8108 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
8110 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
8112 * gst/rtsp-server/rtsp-media.c:
8113 * gst/rtsp-server/rtsp-stream.c:
8114 Revert "rtsp-server: support build against last stable release"
8115 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
8116 Let us require 1.2.3 now, which is going to be released in a few
8119 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
8121 * gst/rtsp-server/rtsp-session-media.c:
8122 * gst/rtsp-server/rtsp-stream-transport.c:
8123 session: improve RTP-Info
8124 Ignore streams that can't generate RTP-Info instead of failing.
8125 Don't return the empty string when all streams are unconfigured but
8126 return NULL so that we don't generate and empty RTP-Info header.
8127 Improve docs a little.
8129 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
8131 * gst/rtsp-server/rtsp-session-media.c:
8132 Don't free rtpinfo GString when it is NULL
8133 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8135 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
8137 * gst/rtsp-server/rtsp-media.c:
8138 media: only set keyframe flag when modifying start
8139 Only set the keyframe flag when we modify the start position. The
8140 keyframe flag should probably be ignored when no change is requested but
8141 until we can claim this is all documented properly and all demuxer
8142 implement this, avoid setting the flag.
8143 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
8145 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
8147 * gst/rtsp-server/rtsp-thread-pool.c:
8148 thread-pool: Unref source after mainloop has quit to avoid races in GLib
8149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
8151 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
8153 * gst/rtsp-server/rtsp-stream.c:
8154 stream: handle NULL seqnum and rtptime arguments
8156 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
8158 * gst/rtsp-server/rtsp-thread-pool.c:
8159 * tests/check/gst/threadpool.c:
8160 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
8161 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
8163 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
8165 * gst/rtsp-server/rtsp-stream.c:
8166 stream: add fallback for missing stats property
8167 Use a fallback when the payloader does not have a stats property
8168 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
8170 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
8173 Automatic update of common submodule
8174 From f7bc1c3 to 1a07da9
8176 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
8178 * gst/rtsp-server/rtsp-stream.c:
8179 stream: don't leak stats structure
8180 Don't leak the stats structure and deal with NULL stats.
8182 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
8184 * gst/rtsp-server/rtsp-stream.c:
8185 stream: Get rtpinfo properties atomically from payloader
8186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
8188 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
8190 * gst/rtsp-server/rtsp-media.c:
8191 media: refactor state change functions and signals
8192 Make functions to set the target state and the pipeline state and emit
8193 the signals from those functions.
8195 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
8197 * gst/rtsp-server/rtsp-media.c:
8198 * gst/rtsp-server/rtsp-media.h:
8199 media: add signal to notify of pending state changes
8201 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
8203 * gst/rtsp-server/rtsp-media.c:
8204 * gst/rtsp-server/rtsp-stream.c:
8205 rtsp-server: support build against last stable release
8206 Until 1.2.3 is out with the new get_type function and we
8209 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
8211 * gst/rtsp-server/rtsp-stream.c:
8212 stream: fix compilation
8214 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
8216 * gst/rtsp-server/rtsp-media.c:
8217 * gst/rtsp-server/rtsp-media.h:
8218 * gst/rtsp-server/rtsp-stream.c:
8219 * gst/rtsp-server/rtsp-stream.h:
8220 stream: add property to configure profiles
8222 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
8224 * gst/rtsp-server/rtsp-client.c:
8225 client: let stream check supported transport
8226 Delegate the check if a transport is allowed to the stream.
8227 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
8229 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
8231 * gst/rtsp-server/rtsp-stream.c:
8232 * gst/rtsp-server/rtsp-stream.h:
8233 stream: add method to check supported transport
8234 Add a method to check if a transport is supported
8236 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
8239 configure.ac: Only check for gstreamer-check, not check
8240 We include check in gstreamer-check since quite some time now.
8242 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
8244 * gst/rtsp-server/rtsp-session-media.c:
8245 * gst/rtsp-server/rtsp-stream-transport.c:
8246 * gst/rtsp-server/rtsp-stream.c:
8247 * gst/rtsp-server/rtsp-stream.h:
8248 stream: return clock-rate from get_rtpinfo
8249 And use it to correct the rtptime to the requested start-time.
8250 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
8252 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
8254 * gst/rtsp-server/rtsp-session-media.c:
8255 * gst/rtsp-server/rtsp-stream-transport.c:
8256 * gst/rtsp-server/rtsp-stream-transport.h:
8257 session-media: calculate start-time
8259 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
8261 * gst/rtsp-server/rtsp-stream-transport.c:
8262 * gst/rtsp-server/rtsp-stream.c:
8263 * gst/rtsp-server/rtsp-stream.h:
8264 stream: also return the running-time
8265 Return the running-time in the rtpinfo as well.
8267 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
8269 * gst/rtsp-server/rtsp-client.c:
8270 * gst/rtsp-server/rtsp-session-media.c:
8271 * gst/rtsp-server/rtsp-session-media.h:
8272 * gst/rtsp-server/rtsp-stream-transport.c:
8273 * gst/rtsp-server/rtsp-stream-transport.h:
8274 session-media: let the session-media make the RTPInfo
8275 Add method to create the RTPInfo for a stream-transport.
8276 Add method to create the RTPInfo for all stream-transports in a
8278 Use the session-media RTPInfo code in client. This allows us to refactor
8279 another method to link the TCP callbacks.
8281 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8283 mount-points: sort sequence before g_sequence_lookup
8284 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
8285 sort sequence if dirty, otherwise lookup will fail.
8286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
8288 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
8291 configure: rename package from gst-rtsp to gst-rtsp-server
8292 To match git module name and avoid confusion with the
8293 rtsp lib in gst-plugins-base and rtsp plugin in -good.
8295 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
8298 configure: bump core/base/good requirement to 1.2.0
8299 Bump to released stable version and make implicit
8300 requirements explicit.
8302 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
8307 Fix broken gettext setup which is not used anyway
8309 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
8312 Automatic update of common submodule
8313 From dbedaa0 to d48bed3
8315 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
8317 * gst/rtsp-server/rtsp-client.c:
8318 * gst/rtsp-server/rtsp-media.c:
8319 * gst/rtsp-server/rtsp-media.h:
8320 media: add setup_sdp vmethod
8321 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
8322 gst_rtsp_media_setup_sdp.
8323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
8325 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
8327 * gst/rtsp-server/rtsp-stream.c:
8328 rtsp-stream: Check return value of sscanf
8329 streamid is only valid if sscanf matched something.
8331 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
8333 * gst/rtsp-server/rtsp-client.c:
8334 rtsp-client: Fix iteration
8335 Wouldn't even enter the code block otherwise (i++ was used as the check
8336 and not the postfix).
8338 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
8340 * gst/rtsp-server/rtsp-client.c:
8341 * gst/rtsp-server/rtsp-client.h:
8342 client: add vmethod to configure media and streams
8343 Implement a vmethod that can be used to configure the media and the
8344 streams based on the current context. Handle the blocksize handling in
8345 the default handler.
8346 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
8348 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8351 Make git ignore more unit test binaries
8353 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
8355 * gst/rtsp-server/rtsp-address-pool.h:
8356 * gst/rtsp-server/rtsp-auth.h:
8357 * gst/rtsp-server/rtsp-client.h:
8358 * gst/rtsp-server/rtsp-context.h:
8359 * gst/rtsp-server/rtsp-media-factory-uri.h:
8360 * gst/rtsp-server/rtsp-media-factory.h:
8361 * gst/rtsp-server/rtsp-media.h:
8362 * gst/rtsp-server/rtsp-mount-points.h:
8363 * gst/rtsp-server/rtsp-server.h:
8364 * gst/rtsp-server/rtsp-session-media.h:
8365 * gst/rtsp-server/rtsp-session-pool.h:
8366 * gst/rtsp-server/rtsp-session.h:
8367 * gst/rtsp-server/rtsp-stream-transport.h:
8368 * gst/rtsp-server/rtsp-stream.h:
8369 * gst/rtsp-server/rtsp-thread-pool.h:
8370 * gst/rtsp-server/rtsp-token.h:
8371 rtsp-server: add padding to many public structures
8372 Not mini objects though, since they are not subclassable
8373 anyway, nor kept on the stack or inlined in a structure.
8375 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
8377 media: add new create_rtpbin vmethod
8378 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
8379 https://bugzilla.gnome.org/show_bug.cgi?id=719734
8381 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
8383 * tests/check/gst/media.c:
8384 tests: fix memory leak, free test's thread pool
8385 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
8387 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
8389 * gst/rtsp-server/rtsp-stream-transport.c:
8390 stream-transport: free url in finalize
8392 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
8394 * gst/rtsp-server/rtsp-media.c:
8395 media: also do state change in suspended state
8397 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
8399 * gst/rtsp-server/rtsp-client.c:
8400 * gst/rtsp-server/rtsp-media.c:
8401 media: also handle prepare and range in suspended state
8402 When we are suspended, we are already prepared.
8403 We can get the range in the suspended state.
8405 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
8407 * tests/check/Makefile.am:
8408 * tests/check/gst/sessionmedia.c:
8409 check: add test for uri in setup
8410 Added unit tests for the new functionality in GstRTSPStreamTransport.
8411 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8413 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
8415 * gst/rtsp-server/rtsp-client.c:
8416 client: store setup uri and use in PLAY response
8417 Store the uri used when doing the setup and use that in the PLAY
8419 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
8421 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
8423 * gst/rtsp-server/rtsp-stream-transport.c:
8424 * gst/rtsp-server/rtsp-stream-transport.h:
8425 stream-transport: add method to get/set url
8427 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
8429 * gst/rtsp-server/rtsp-client.c:
8430 client: suspend after SDP and unsuspend before PLAYING
8431 Based on patches by Ognyan Tonchev <ognyan@axis.com>
8432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
8434 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
8436 * gst/rtsp-server/rtsp-media-factory.c:
8437 * gst/rtsp-server/rtsp-media-factory.h:
8438 * gst/rtsp-server/rtsp-media.c:
8439 * gst/rtsp-server/rtsp-media.h:
8440 * gst/rtsp-server/rtsp-session-media.c:
8441 * gst/rtsp-server/rtsp-session.c:
8442 * tests/check/gst/media.c:
8443 * tests/check/gst/mediafactory.c:
8444 media: add suspend modes
8445 Add support for different suspend modes. The stream is suspended right after
8446 producing the SDP and after PAUSE. Different suspend modes are available that
8447 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
8448 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
8449 state and RESET will bring the pipeline to the NULL state.
8450 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
8451 this means that the pipeline needs to be prerolled again.
8452 Base on patches by Ognyan Tonchev <ognyan@axis.com>
8453 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8455 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
8457 * gst/rtsp-server/rtsp-media.c:
8458 media: start live streams in blocked state
8459 Start live streams in the blocked state and make them preroll using the
8460 messages. This ensure that no data is played by the sink until we explicitly
8461 unblock the stream right before going to PLAYING.
8462 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8464 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
8466 * gst/rtsp-server/rtsp-media.c:
8467 media: refactor starting and waiting for preroll
8468 Based on patches from Ognyan Tonchev <ognyan@axis.com>
8469 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8471 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
8473 * gst/rtsp-server/rtsp-stream.c:
8474 * gst/rtsp-server/rtsp-stream.h:
8475 stream: add API to block streams
8476 Add an API to block on the streams and make it post a message.
8477 Based on patch by Ognyan Tonchev <ognyan@axis.com>
8478 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
8480 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
8482 * docs/libs/Makefile.am:
8483 docs: Specify the override file
8484 Even if it's empty (for now) it avoids make distcheck complaining
8486 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
8488 * gst/rtsp-server/rtsp-media.c:
8489 media: move default implementations to where they are used
8491 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
8493 * gst/rtsp-server/rtsp-media.c:
8494 media: take the right lock in gst_rtsp_media_set_pipeline_state()
8495 We need to take the state_lock when calling this method.
8497 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
8499 * gst/rtsp-server/rtsp-media.c:
8500 media: handle add-added on non-bins too
8501 Handle dynamic payloaders that are not bins, as used in the unit-test.
8503 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8505 * gst/rtsp-server/rtsp-media-factory.c:
8506 * gst/rtsp-server/rtsp-media-factory.h:
8507 * gst/rtsp-server/rtsp-media.c:
8508 rtsp-media/-factory: Fix request pad name comments
8509 These must be escaped for gtk-doc to parse the comments without warnings.
8511 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8513 rtsp-media: remove transports if media is in error status
8514 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
8515 trying to change to GST_STATE_NULL and media is in error status, we
8516 remove all transports.
8517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
8519 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
8521 * gst/rtsp-server/rtsp-media.c:
8522 rtsp-media: use element metadata to find payloader
8523 Use the element metadata to find the payloader instead of checking
8525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
8527 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
8529 rtsp-stream: add getter for payload type
8530 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
8531 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
8532 element and create the stream with this one instead of the dynpay%d
8534 https://bugzilla.gnome.org/show_bug.cgi?id=712396
8536 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8538 * gst/rtsp-server/rtsp-client.c:
8539 * gst/rtsp-server/rtsp-context.h:
8540 * gst/rtsp-server/rtsp-media.c:
8541 * gst/rtsp-server/rtsp-mount-points.c:
8542 * gst/rtsp-server/rtsp-server.c:
8543 * gst/rtsp-server/rtsp-token.c:
8544 rtsp-*: Refer to NULL as a constant in comments
8546 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8548 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8550 rtsp-*: Fix type name typos in comments
8551 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
8552 * rtsp-auth: Refer to part of constant name as text
8553 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
8554 * rtsp-session-media: Fix GstRTSPSessionMedia typo
8555 * rtsp-stream: Fix typo when refering to GstBin
8556 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8558 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8561 * docs/libs/gst-rtsp-server-docs.sgml:
8562 * docs/libs/gst-rtsp-server-sections.txt:
8563 docs: Improve documentation
8564 * Include annotation-glossary to quiet gtk-doc
8565 * Rename remaining ClientState -> Context
8566 * Rename object hierarchy file
8567 * Remove stale chapter references
8568 * Add missing function and object references
8569 * Include missing GstRTSPAddressPoolResult
8570 https://bugzilla.gnome.org/show_bug.cgi?id=714988
8572 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
8574 * gst/rtsp-server/rtsp-client.c:
8575 * gst/rtsp-server/rtsp-server.c:
8576 * gst/rtsp-server/rtsp-session-pool.c:
8577 * gst/rtsp-server/rtsp-session.c:
8578 * gst/rtsp-server/rtsp-stream.c:
8579 rtsp-server: sprinkle some allow-none annotations for g-i
8581 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
8583 * gst/rtsp-server/rtsp-stream.c:
8584 * gst/rtsp-server/rtsp-stream.h:
8585 stream: add method to filter transports
8586 Add a method to safely iterate and collect the stream transports
8587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
8589 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
8591 * gst/rtsp-server/rtsp-client.c:
8592 * gst/rtsp-server/rtsp-server.c:
8593 * gst/rtsp-server/rtsp-session-pool.c:
8594 * gst/rtsp-server/rtsp-session.c:
8595 rtsp: allow NULL func in filters
8596 Passing a null function make the filters return a list of
8599 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8601 * gst/rtsp-server/rtsp-address-pool.c:
8602 * tests/check/gst/addresspool.c:
8603 address-pool: fix address increment
8604 Use a guint instead of guint8 to increment the address. It's still not
8605 completely correct because a guint might not be able to hold the complete
8606 address range, but that's an enhacement for later.
8607 Add unit test to test improved behaviour.
8608 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8610 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8612 * gst/rtsp-server/rtsp-client.c:
8613 * tests/check/gst/client.c:
8614 client: allow absolute path in requests
8615 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8617 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8619 * gst/rtsp-server/rtsp-client.c:
8620 * gst/rtsp-server/rtsp-client.h:
8621 client: make make_path_from_uri a vmethod
8623 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8625 * docs/libs/gst-rtsp-server-sections.txt:
8626 * gst/rtsp-server/rtsp-stream.c:
8627 * gst/rtsp-server/rtsp-stream.h:
8628 * tests/check/Makefile.am:
8629 * tests/check/gst/stream.c:
8630 stream: Add functions to get rtp and rtcp sockets
8631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8633 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8635 * gst/rtsp-server/rtsp-context.c:
8636 * gst/rtsp-server/rtsp-context.h:
8637 context: defing a GType for the context
8638 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8640 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8642 * gst/rtsp-server/Makefile.am:
8643 * gst/rtsp-server/rtsp-auth.c:
8644 * gst/rtsp-server/rtsp-context.c:
8645 * gst/rtsp-server/rtsp-media.c:
8646 * gst/rtsp-server/rtsp-mount-points.c:
8647 * gst/rtsp-server/rtsp-server.h:
8648 * gst/rtsp-server/rtsp-session-media.c:
8649 * gst/rtsp-server/rtsp-session.c:
8650 * gst/rtsp-server/rtsp-stream.c:
8651 Fixed several GIR warnings
8653 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8655 * gst/rtsp-server/rtsp-auth.c:
8658 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8660 * tests/check/Makefile.am:
8661 * tests/check/gst/token.c:
8662 tests: Add unit tests for token
8663 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8665 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8667 * gst/rtsp-server/rtsp-token.c:
8668 token: Validate args for gst_rtsp_token_is_allowed
8669 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8671 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8673 * gst/rtsp-server/rtsp-token.c:
8674 token: Fix bug when creating empty token
8675 We always want to have a valid GstStructure in the token.
8676 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8678 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8680 * gst/rtsp-server/rtsp-thread-pool.c:
8681 thread-pool: avoid race in shutdown
8682 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8683 don't actually stop the mainloop ever. Solve this race by adding an idle source
8684 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8685 if quit was called before we started it.
8687 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8689 * tests/check/Makefile.am:
8690 * tests/check/gst/permissions.c:
8691 tests: Add unit tests for permissions
8692 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8694 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8696 * tests/check/gst/mediafactory.c:
8697 tests: Test mediafactory permissions
8698 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8700 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8702 * gst/rtsp-server/rtsp-permissions.c:
8703 permissions: Fix refcounting when adding/removing roles
8704 Previously a role that was removed was unreffed twice, and when
8705 replacing an existing role the replaced role was freed while still being
8706 referenced. Both bugs are now fixed.
8707 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8709 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8711 * tests/check/gst/media.c:
8712 * tests/check/gst/mediafactory.c:
8713 * tests/check/gst/rtspserver.c:
8714 tests: Check gst_rtsp_url_parse return value
8715 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8717 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8720 Automatic update of common submodule
8721 From 865aa20 to dbedaa0
8723 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8725 * gst/rtsp-server/rtsp-server.c:
8726 rtsp-server: Fix socket leak
8727 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8729 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8731 * gst/rtsp-server/rtsp-session-pool.c:
8732 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8733 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8735 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8737 * examples/.gitignore:
8738 * examples/test-video.c:
8739 examples: fix compilation when WITH_AUTH is defined
8740 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8742 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8745 gitignore: Add new test binary
8747 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8749 * tests/check/Makefile.am:
8750 * tests/check/gst/threadpool.c:
8751 thread-pool: Add unit test for the thread pools
8752 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8754 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8756 * gst/rtsp-server/rtsp-thread-pool.c:
8757 thread-pool: Fix thread leak when reusing threads
8758 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8760 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8762 * gst/rtsp-server/rtsp-server.c:
8763 * tests/check/gst/rtspserver.c:
8764 tests: fixed racy behavior in rtspserver tests
8765 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8767 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8769 * tests/check/gst/addresspool.c:
8770 tests: Improve address pool unit tests
8771 Add a range with mixed IPV4 and IPV6 addresses to pool.
8772 Get an IPV4 address from an IPV6-only pool.
8773 Get an IPV6 address from an IPV4-only pool.
8774 Reserve a IPV6 address from an IPV4-only pool.
8775 Check for unicast addresses in multicast-only pool.
8776 Check for unicast addresses in uni-/multicast-mixed pool.
8777 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8779 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8781 * gst/rtsp-server/rtsp-client.c:
8782 client: append query string in PAUSE/PLAY/TEARDOWN as well
8784 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8786 * gst/rtsp-server/rtsp-client.c:
8787 client: Add query to control path
8788 If the SETUP url contains a query it must be appended to the control
8789 path so that it matches any already created stream in the media. The
8790 query will also be appended to the session media path.
8792 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8794 * gst/rtsp-server/rtsp-media.c:
8795 rtsp-media: remove old line
8797 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8799 * gst/rtsp-server/rtsp-stream.c:
8800 stream: Correct control comparison
8801 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8803 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8805 * gst/rtsp-server/rtsp-media.c:
8806 media: Check dynamically if the pipeline supports seeking
8807 We should not depend on whether or not the pipeline state change
8808 returned NO_PREROLL or not. A media could dynamically change its
8809 element and switch from seekable to non seekable so it's best to test
8810 the seekable nature of the pipeline dynamically when we try to do a seek.
8812 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8814 * gst/rtsp-server/rtsp-media.c:
8815 media: Return FALSE if seeking is not supported
8817 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8819 * gst/rtsp-server/rtsp-media.c:
8820 rtsp-media: don't seek accurate by default
8821 Accurate seeking is perhaps a little overkill in the most common situation and
8822 causes some formats (mp3) over slow media to seek extremely slowly.
8824 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8826 * tests/check/gst/rtspserver.c:
8827 tests: fix unit test
8828 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8830 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8832 * gst/rtsp-server/rtsp-client.c:
8833 client: Reply 400 if media cannot be constructed
8834 Reply 400 Bad Request instead of 503 Service Unavailable if media
8835 cannot be constructed in SETUP.
8836 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8838 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8840 * gst/rtsp-server/rtsp-client.c:
8841 client: Send setup reply once only
8842 If find_media() failed in handle_setup_request() two replies was sent.
8843 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8845 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8848 Automatic update of common submodule
8849 From 6b03ba7 to 865aa20
8851 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8853 * gst/rtsp-server/rtsp-server.c:
8854 server: Emit client-connected signal earlier
8855 Emit client-connected before the client ref is given to a GSource,
8856 otherwise client-connected can be emitted after the client object has
8859 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8861 * gst/rtsp-server/rtsp-address-pool.c:
8862 * gst/rtsp-server/rtsp-address-pool.h:
8863 * gst/rtsp-server/rtsp-stream.c:
8864 * tests/check/gst/addresspool.c:
8865 addresspool: return reason of failure
8866 Let gst_rtsp_address_pool_reserve_address() return the reason why
8867 the address could not be reserved.
8868 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8870 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8873 autogen.sh: Sync behaviour with other GStreamer modules
8874 Allows building from outside of tree amongst other things
8876 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8879 Automatic update of common submodule
8880 From b613661 to 6b03ba7
8882 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8885 Automatic update of common submodule
8886 From 74a6857 to b613661
8888 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8891 Automatic update of common submodule
8892 From 01a7a46 to 74a6857
8894 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8896 * gst/rtsp-server/rtsp-client.c:
8897 client: Do not read beyond end of path string
8898 If the setup was done without a control url, make sure we don't try to read the
8899 non-existing control string and crash.
8901 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8903 * gst/rtsp-server/rtsp-client.c:
8904 client: Fix RTPInfo header
8905 Refactor the method to make the content_base.
8906 Use the content-base and the control url to construct the RTPInfo
8909 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8911 * gst/rtsp-server/rtsp-client.c:
8912 client: map url to path only in describe
8913 Only map the request url to a path in the DESCRIBE method. The SDP then
8914 contains the base and control urls that should be used to SETUP/PAUSE/
8915 PLAY/TEARDOWN the media.
8917 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-client.c:
8920 Revert "client: map URL to path in requests"
8921 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8922 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8923 contains the base and control urls which are used in the SETUP, PLAY,
8924 PAUSE and TEARDOWN requests.
8926 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-client.c:
8929 client: map URL to path in requests
8931 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8933 * gst/rtsp-server/rtsp-client.c:
8934 * gst/rtsp-server/rtsp-mount-points.c:
8935 * gst/rtsp-server/rtsp-mount-points.h:
8936 mount-points: make vmethod to make path from uri
8937 Make a vmethod to transform an url into a path. The path is then used to lookup
8938 the factory. This makes it possible to also use other bits of the url, such as
8939 the query parameters, to locate the factory.
8941 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8943 * gst/rtsp-server/rtsp-thread-pool.c:
8944 * gst/rtsp-server/rtsp-thread-pool.h:
8945 thread-pool: Add cleanup to wait for the threadpool to finish
8946 Also fix race condition if two threads are asking for the first
8947 thread from the thread pool at once. This would case two internal
8948 GThreadPools to be created.
8949 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8951 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8953 * gst/rtsp-server/rtsp-client.c:
8954 * tests/check/gst/client.c:
8955 client: free threadpool
8956 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8958 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8960 * tests/check/gst/mountpoints.c:
8961 mountpoints tests: unref matched factories
8962 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8964 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8966 * tests/check/gst/media.c:
8967 media tests: unref thread pool and caps
8968 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8970 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8972 * gst/rtsp-server/rtsp-auth.c:
8973 * gst/rtsp-server/rtsp-media-factory.c:
8974 * gst/rtsp-server/rtsp-media.c:
8975 auth, media, media-factory: unref permissions
8976 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8978 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8980 * examples/Makefile.am:
8981 Makefile: add rule for appsrc example
8983 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8985 * examples/test-appsrc.c:
8986 tests: add appsrc example
8987 Add an example on how to use appsrc to feed the server pipeline with data.
8989 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8991 * gst/rtsp-server/rtsp-client.c:
8992 rtsp-client: remove query part from content-base string
8993 Make sure that after the control url has been resolved, it's
8994 not a part of the query-string.
8995 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8997 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8999 * gst/rtsp-server/rtsp-client.c:
9000 client: don't check url in response
9001 There is no url or method in the response to check
9003 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9005 * gst/rtsp-server/rtsp-client.c:
9006 * gst/rtsp-server/rtsp-client.h:
9007 Add handle-response signal for when we receive a GET_PARAMETER response
9009 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9011 * gst/rtsp-server/rtsp-server.c:
9012 Fix gst_rtsp_server_client_filter, using wrong variable type
9014 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
9016 * gst/rtsp-server/rtsp-media-factory-uri.c:
9017 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
9018 For AAC we need to check for framed=true instead of parsed=true.
9019 https://bugzilla.gnome.org/show_bug.cgi?id=701384
9021 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9023 * gst/rtsp-server/rtsp-stream.c:
9024 stream: optimize pipeline for protocols
9025 When TCP is not an allowed protocol for the stream, avoid creating the
9026 appsrc/appsink/queue and tee elements.
9028 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9030 * gst/rtsp-server/rtsp-media.c:
9031 media: set protocols on streams
9033 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9035 * gst/rtsp-server/rtsp-client.c:
9036 client: use protocols supported by stream
9038 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9040 * gst/rtsp-server/rtsp-media-factory.c:
9041 * gst/rtsp-server/rtsp-media.c:
9042 * gst/rtsp-server/rtsp-stream.c:
9043 media-factory: allow all protocols
9045 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9047 * gst/rtsp-server/rtsp-media.c:
9048 media: configure protocols in new streams
9050 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9052 * gst/rtsp-server/rtsp-stream.c:
9053 * gst/rtsp-server/rtsp-stream.h:
9054 stream: add protocols property
9056 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9058 * gst/rtsp-server/rtsp-media.c:
9059 rtsp-media: send state in "new-state" signal
9060 https://bugzilla.gnome.org/show_bug.cgi?id=705110
9062 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
9065 build: add subdir-objects to AM_INIT_AUTOMAKE
9066 Fixes warnings with automake 1.14
9067 https://bugzilla.gnome.org/show_bug.cgi?id=705350
9069 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9071 * docs/libs/gst-rtsp-server-sections.txt:
9072 * gst/rtsp-server/rtsp-client.c:
9073 * gst/rtsp-server/rtsp-server.c:
9074 * gst/rtsp-server/rtsp-server.h:
9075 server: add method to iterate clients of server
9077 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9079 * gst/rtsp-server/rtsp-media.c:
9080 * gst/rtsp-server/rtsp-media.h:
9081 Add vmethod for rtsp-media subclass to access rtpbin
9083 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9085 * gst/rtsp-server/rtsp-client.h:
9086 small documentation fix
9088 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9090 * gst/rtsp-server/rtsp-client.c:
9091 Do not take range header if range is invalid
9093 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9095 * docs/libs/gst-rtsp-server-sections.txt:
9096 * gst/rtsp-server/rtsp-media.c:
9097 media: add docs for new method
9099 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9101 * gst/rtsp-server/rtsp-media.c:
9102 * gst/rtsp-server/rtsp-media.h:
9103 Add API to rtsp-media set the pipeline's state
9105 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-media.c:
9108 Update current position/duration when gst_rtsp_media_get_range_string is called
9110 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * examples/test-cgroups.c:
9113 tests: add some more docs
9115 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9117 * examples/test-cgroups.c:
9118 * gst/rtsp-server/Makefile.am:
9119 * gst/rtsp-server/rtsp-auth.c:
9120 * gst/rtsp-server/rtsp-auth.h:
9121 * gst/rtsp-server/rtsp-client.c:
9122 * gst/rtsp-server/rtsp-client.h:
9123 * gst/rtsp-server/rtsp-context.c:
9124 * gst/rtsp-server/rtsp-context.h:
9125 * gst/rtsp-server/rtsp-params.c:
9126 * gst/rtsp-server/rtsp-params.h:
9127 * gst/rtsp-server/rtsp-server.c:
9128 * gst/rtsp-server/rtsp-thread-pool.c:
9129 * gst/rtsp-server/rtsp-thread-pool.h:
9130 * tests/check/gst/client.c:
9131 ClientState -> Context
9132 Rename the clientstate to context and put the code in a separate file.
9134 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9136 * examples/test-auth.c:
9137 * gst/rtsp-server/rtsp-auth.c:
9138 * gst/rtsp-server/rtsp-auth.h:
9139 auth: add support for default token
9140 The default token is used when the user is not authenticated and can be used to
9141 give minimal permissions.
9143 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9145 * examples/test-auth.c:
9146 * gst/rtsp-server/rtsp-auth.c:
9147 auth: use defines when possible
9149 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9151 * gst/rtsp-server/rtsp-address-pool.c:
9152 address-pool: improve docs
9154 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9156 * gst/rtsp-server/rtsp-permissions.c:
9157 permissions: add the role to the copy
9159 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
9161 * gst/rtsp-server/rtsp-permissions.c:
9162 permissions: Also copy the roles
9164 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
9166 * gst/rtsp-server/rtsp-permissions.c:
9167 permissions: Make it build
9169 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9171 * gst/rtsp-server/rtsp-address-pool.h:
9174 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9176 * docs/libs/gst-rtsp-server-sections.txt:
9177 * gst/rtsp-server/rtsp-auth.c:
9178 * gst/rtsp-server/rtsp-auth.h:
9179 * gst/rtsp-server/rtsp-media.c:
9180 * gst/rtsp-server/rtsp-session-media.c:
9181 * gst/rtsp-server/rtsp-stream-transport.c:
9182 * gst/rtsp-server/rtsp-stream-transport.h:
9183 * gst/rtsp-server/rtsp-stream.c:
9184 * tests/check/gst/client.c:
9187 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9189 * docs/libs/gst-rtsp-server-sections.txt:
9190 * gst/rtsp-server/rtsp-address-pool.c:
9191 * gst/rtsp-server/rtsp-address-pool.h:
9192 * tests/check/gst/addresspool.c:
9193 * tests/check/gst/rtspserver.c:
9194 address-pool: cleanups
9195 Remove redundant method, improve docs.
9197 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9199 * docs/libs/gst-rtsp-server-sections.txt:
9200 * gst/rtsp-server/rtsp-auth.h:
9201 * gst/rtsp-server/rtsp-permissions.c:
9202 * gst/rtsp-server/rtsp-permissions.h:
9203 * gst/rtsp-server/rtsp-token.c:
9206 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9208 * gst/rtsp-server/rtsp-permissions.c:
9209 permissions: implement _remove_role
9211 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9213 * gst/rtsp-server/rtsp-permissions.c:
9214 permissions: update docs
9216 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9218 * tests/check/gst/client.c:
9219 tests: simplify tests
9220 Client settings are now disabled by default so we don't need an auth
9221 module to disable them.
9223 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9225 * gst/rtsp-server/rtsp-auth.c:
9226 auth: add default authorizations
9227 When no auth module is specified, use our table of defaults to look up the
9228 default value of the check instead of always allowing everything. This was
9229 we can disallow client settings by default.
9231 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9234 README: update readme
9236 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9238 * gst/rtsp-server/rtsp-thread-pool.c:
9239 * gst/rtsp-server/rtsp-thread-pool.h:
9240 thread-pool: add more docs
9242 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9244 * gst/rtsp-server/rtsp-thread-pool.c:
9245 * gst/rtsp-server/rtsp-thread-pool.h:
9246 thread-pool: fix race in thread reuse
9247 If we try to reuse a thread right after we made it stop, we end up using a
9248 stopped thread. Catch this case and only reuse threads that are not stopping.
9250 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9252 * gst/rtsp-server/rtsp-server.c:
9253 server: add small debug
9255 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9257 * tests/check/gst/client.c:
9259 Add some permissions to media so we can use the auth and enable
9262 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9264 * gst/rtsp-server/rtsp-client.c:
9265 client: support pushed context in handle_request
9266 If we already have a pushed state, reuse it and add our own things. This makes
9267 it easier to write tests.
9269 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9271 * gst/rtsp-server/rtsp-auth.c:
9272 auth: don't auth on methods
9273 Don't authorize on methods anymore but on the resources that we
9274 try to access, this is more flexible.
9275 Move the authorization checks to where they are needed and let the
9276 check return the response on error.
9278 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9280 * gst/rtsp-server/rtsp-mount-points.c:
9281 mount-points: add some debug
9283 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9285 * tests/check/gst/client.c:
9286 tests: almost fix test
9288 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9290 * gst/rtsp-server/rtsp-auth.c:
9291 * gst/rtsp-server/rtsp-auth.h:
9292 * gst/rtsp-server/rtsp-client.c:
9293 * gst/rtsp-server/rtsp-client.h:
9294 * gst/rtsp-server/rtsp-server.c:
9295 * gst/rtsp-server/rtsp-server.h:
9296 auth: let the auth module check client_settings
9297 Let the auth module decide if client settings are allowed for the
9300 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9302 * gst/rtsp-server/rtsp-token.c:
9303 * gst/rtsp-server/rtsp-token.h:
9304 token: add method to check boolean permission
9306 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * examples/test-auth.c:
9309 * examples/test-cgroups.c:
9310 * gst/rtsp-server/rtsp-token.c:
9311 * gst/rtsp-server/rtsp-token.h:
9312 token: simplify token constructor
9313 Use variable arguments to make easier API.
9315 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9317 * examples/test-auth.c:
9318 * examples/test-cgroups.c:
9319 * gst/rtsp-server/rtsp-media-factory.c:
9320 * gst/rtsp-server/rtsp-media-factory.h:
9321 media-factory: add convenience API for factory
9323 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9325 * examples/test-auth.c:
9326 * examples/test-cgroups.c:
9327 * gst/rtsp-server/rtsp-permissions.c:
9328 * gst/rtsp-server/rtsp-permissions.h:
9329 permissions: simplify API a little
9330 Avoid passing GstStructure in the add_role method, use varargs instead
9331 to construct the structure behind the scenes. We can then also use the
9332 structure name as the role and simplify some more logic.
9334 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9336 * gst/rtsp-server/rtsp-auth.c:
9339 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9341 * gst/rtsp-server/rtsp-auth.c:
9342 * gst/rtsp-server/rtsp-auth.h:
9343 * gst/rtsp-server/rtsp-client.c:
9344 auth: handle unauthorized response
9345 Move handling of the unauthorized response to the auth module, it can add
9346 the appropriate headers to request authorization for the required method
9347 much better than the client.
9349 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9351 * gst/rtsp-server/rtsp-client.c:
9352 * gst/rtsp-server/rtsp-client.h:
9353 client: allow for sending any message, not only requests
9354 Change the _send_request() method to _send_message() so that we
9355 can both send requests and replies.
9357 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9359 * docs/libs/gst-rtsp-server-sections.txt:
9360 * gst/rtsp-server/rtsp-server.h:
9363 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9365 * examples/test-video.c:
9366 * gst/rtsp-server/rtsp-auth.c:
9367 * gst/rtsp-server/rtsp-auth.h:
9368 * gst/rtsp-server/rtsp-server.c:
9369 * gst/rtsp-server/rtsp-server.h:
9370 auth: move TLS handling to auth module
9371 Remove the TLS settings on the server and move it to the auth module because
9372 that is where security related bits go.
9374 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9376 * gst/rtsp-server/rtsp-client.c:
9377 * gst/rtsp-server/rtsp-client.h:
9378 client: add state push/pop
9380 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9382 * gst/rtsp-server/rtsp-client.c:
9383 * gst/rtsp-server/rtsp-client.h:
9384 client: add connection to state
9386 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9388 * gst/rtsp-server/rtsp-mount-points.c:
9389 mount-points: fix debug
9391 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9393 * tests/check/gst/media.c:
9394 tests: fix media test
9396 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9398 * gst/rtsp-server/rtsp-thread-pool.c:
9399 thread-pool: we don't require a state
9401 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9403 * gst/rtsp-server/rtsp-server.c:
9404 server: let context ref the server
9405 So that we don't risk losing the server object early anc crash.
9407 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9409 * tests/check/gst/client.c:
9410 tests: fix client test
9412 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9415 * docs/libs/gst-rtsp-server-docs.sgml:
9416 * docs/libs/gst-rtsp-server-sections.txt:
9417 * gst/rtsp-server/rtsp-address-pool.c:
9418 * gst/rtsp-server/rtsp-auth.c:
9419 * gst/rtsp-server/rtsp-client.c:
9420 * gst/rtsp-server/rtsp-client.h:
9421 * gst/rtsp-server/rtsp-media-factory-uri.c:
9422 * gst/rtsp-server/rtsp-media-factory.c:
9423 * gst/rtsp-server/rtsp-media-factory.h:
9424 * gst/rtsp-server/rtsp-media.c:
9425 * gst/rtsp-server/rtsp-mount-points.c:
9426 * gst/rtsp-server/rtsp-params.c:
9427 * gst/rtsp-server/rtsp-permissions.c:
9428 * gst/rtsp-server/rtsp-sdp.c:
9429 * gst/rtsp-server/rtsp-server.c:
9430 * gst/rtsp-server/rtsp-server.h:
9431 * gst/rtsp-server/rtsp-session-media.c:
9432 * gst/rtsp-server/rtsp-session-pool.c:
9433 * gst/rtsp-server/rtsp-session.c:
9434 * gst/rtsp-server/rtsp-stream-transport.c:
9435 * gst/rtsp-server/rtsp-stream.c:
9436 * gst/rtsp-server/rtsp-thread-pool.c:
9437 * gst/rtsp-server/rtsp-token.c:
9440 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * gst/rtsp-server/rtsp-session-pool.c:
9443 * gst/rtsp-server/rtsp-session-pool.h:
9444 session-pool: make vmethod to create a session
9445 Make a vmethod to create a sessions so that subclasses can create
9446 custom session objects
9448 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9450 * gst/rtsp-server/rtsp-auth.c:
9451 * gst/rtsp-server/rtsp-media-factory.h:
9452 * gst/rtsp-server/rtsp-media.h:
9453 * gst/rtsp-server/rtsp-mount-points.h:
9454 * gst/rtsp-server/rtsp-session-pool.h:
9455 * gst/rtsp-server/rtsp-stream.h:
9458 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9460 * docs/libs/gst-rtsp-server-docs.sgml:
9461 * docs/libs/gst-rtsp-server-sections.txt:
9462 * gst/rtsp-server/rtsp-address-pool.c:
9463 * gst/rtsp-server/rtsp-address-pool.h:
9464 * gst/rtsp-server/rtsp-auth.c:
9465 * gst/rtsp-server/rtsp-client.h:
9466 * gst/rtsp-server/rtsp-media-factory.h:
9467 * gst/rtsp-server/rtsp-media.c:
9468 * gst/rtsp-server/rtsp-media.h:
9469 * gst/rtsp-server/rtsp-permissions.c:
9470 * gst/rtsp-server/rtsp-permissions.h:
9471 * gst/rtsp-server/rtsp-server.h:
9472 * gst/rtsp-server/rtsp-session-media.c:
9473 * gst/rtsp-server/rtsp-session-media.h:
9474 * gst/rtsp-server/rtsp-session-pool.h:
9475 * gst/rtsp-server/rtsp-session.h:
9476 * gst/rtsp-server/rtsp-stream-transport.h:
9477 * gst/rtsp-server/rtsp-stream.c:
9478 * gst/rtsp-server/rtsp-thread-pool.h:
9481 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9484 * examples/Makefile.am:
9485 configure: compile cgroup example conditionally
9486 Only compile the cgroup example when we have libcgroup
9488 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9491 * examples/Makefile.am:
9492 * examples/test-cgroups.c:
9493 examples: add cgroups example
9495 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9497 * tests/check/gst/rtspserver.c:
9498 tests: fix compilation
9500 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9502 * gst/rtsp-server/rtsp-thread-pool.c:
9503 thread-pool: fix vmethod invocation
9505 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9507 * gst/rtsp-server/rtsp-thread-pool.c:
9508 * gst/rtsp-server/rtsp-thread-pool.h:
9509 thread-pool: store thread type in thread
9511 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9513 * gst/rtsp-server/rtsp-client.c:
9514 client: pass thread from pool to media _prepare
9515 Get a thread from the configured threadpool and pass it to the prepare method of
9518 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9520 * gst/rtsp-server/rtsp-media.c:
9521 * gst/rtsp-server/rtsp-media.h:
9522 media: Accept a thread in _prepare
9523 Remove out own threadpool handling and use the provided thread and
9524 maincontext for the bus messages and the state changes.
9526 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9528 * gst/rtsp-server/rtsp-server.c:
9529 server: configure client thread pool
9531 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9533 * gst/rtsp-server/rtsp-client.c:
9534 * gst/rtsp-server/rtsp-client.h:
9535 client: add method to configure thread pool
9537 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9539 * gst/rtsp-server/rtsp-client.h:
9540 * gst/rtsp-server/rtsp-server.c:
9541 * gst/rtsp-server/rtsp-server.h:
9542 server: use thread pool
9543 Use the thread pool instead of doing our own thing.
9545 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9547 * gst/rtsp-server/Makefile.am:
9548 * gst/rtsp-server/rtsp-thread-pool.c:
9549 * gst/rtsp-server/rtsp-thread-pool.h:
9550 thread-pool: add object to manage threads
9551 Add an object to manage the client and media threads.
9553 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9555 * gst/rtsp-server/rtsp-auth.c:
9556 auth: debug authorization check
9558 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9560 * gst/rtsp-server/rtsp-media.c:
9561 media: start media pipeline in context
9562 Start the media pipeline in the provided context (or our default one
9563 when NULL). This makes sure that we run the bus thread in this context and that
9564 all media threads are children of this context.
9566 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9568 * gst/rtsp-server/rtsp-media-factory.c:
9569 factory: pass permissions to media by default
9571 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9573 * examples/test-auth.c:
9574 test: add permissions to auth test
9575 Ass some permissions to the media factory in the test.
9577 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9579 * gst/rtsp-server/rtsp-auth.c:
9580 * gst/rtsp-server/rtsp-auth.h:
9581 * gst/rtsp-server/rtsp-client.c:
9582 auth: simplify auth checks
9583 Remove client from methods, it's now in the state
9584 Perform the check specified by the string, use the information from the
9585 thread local context.
9587 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9589 * gst/rtsp-server/rtsp-client.c:
9590 * gst/rtsp-server/rtsp-client.h:
9591 client: add state to current thread
9592 Add the client to the ClientState object.
9593 Place the ClientState on the current thread.
9595 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9597 * gst/rtsp-server/rtsp-media-factory.c:
9598 * gst/rtsp-server/rtsp-media-factory.h:
9599 * gst/rtsp-server/rtsp-media.c:
9600 * gst/rtsp-server/rtsp-media.h:
9601 media: make it possible to set permissions
9602 Make it possible to set permissions on media and media factory objects
9604 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9606 * gst/rtsp-server/Makefile.am:
9607 * gst/rtsp-server/rtsp-permissions.c:
9608 * gst/rtsp-server/rtsp-permissions.h:
9609 permissions: add permissions object
9610 Add a mini object to store permissions based on a role.
9612 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9614 * examples/test-auth.c:
9615 * gst/rtsp-server/rtsp-auth.c:
9616 * gst/rtsp-server/rtsp-auth.h:
9617 * gst/rtsp-server/rtsp-client.c:
9618 auth: add auth checks
9619 Add an enum with auth checks and implement the checks in the auth object.
9620 Perform the checks from the client.
9622 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9624 * examples/test-auth.c:
9625 * gst/rtsp-server/rtsp-auth.c:
9626 * gst/rtsp-server/rtsp-auth.h:
9627 * gst/rtsp-server/rtsp-client.h:
9628 auth: use the token after authentication
9629 After we authenticated a user, keep the Token around in the state.
9631 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9633 * gst/rtsp-server/rtsp-client.c:
9634 * gst/rtsp-server/rtsp-media.c:
9635 * gst/rtsp-server/rtsp-media.h:
9636 * tests/check/gst/media.c:
9637 media: add optional context for bus messages
9638 Add an optional mainloop to _prepare that will handle the bus messages instead
9639 of always using the shared mainloop.
9641 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9643 * gst/rtsp-server/Makefile.am:
9644 * gst/rtsp-server/rtsp-token.c:
9645 * gst/rtsp-server/rtsp-token.h:
9646 token: add authorization token
9647 Add a simply miniobject that contains the authorizations. The object contains a
9648 GstStructure that hold all authorization fields. When a user is authenticated,
9649 the auth module will create a Token for the user. The token is then used to
9650 check what operations the user is allowed to do and various other configuration
9653 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9655 * examples/test-auth.c:
9656 * gst/rtsp-server/rtsp-auth.c:
9657 * gst/rtsp-server/rtsp-auth.h:
9658 * gst/rtsp-server/rtsp-client.c:
9659 * gst/rtsp-server/rtsp-client.h:
9660 * gst/rtsp-server/rtsp-media-factory.c:
9661 * gst/rtsp-server/rtsp-media-factory.h:
9662 * gst/rtsp-server/rtsp-media.c:
9663 * gst/rtsp-server/rtsp-media.h:
9664 auth: remove auth from media and factory
9665 Remove the auth object from media and factory. We want to have the RTSPClient
9666 authenticate and authorize resources, there is no need to place another auth
9667 manager on the media/factory.
9669 2013-07-04 14:33:59 +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.h:
9675 auth: add support for multiple basic auth tokens
9676 Make it possible to add multiple basic authorisation tokens to one authorization
9677 object. Associate with each token an authorization group that will define what
9678 capabilities are allowed.
9680 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9682 * gst/rtsp-server/rtsp-client.c:
9683 client: error out on non-aggregate control
9684 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9686 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9688 * gst/rtsp-server/rtsp-client.c:
9689 client: rework setup request a little
9690 Cache the media in DESCRIBE based on the longest matching path with the uri
9691 that we can find in the mount points.
9692 Rework the setup request a little to get the media from the session or from
9693 the longest matching path, this way we can derive the control string as
9694 everything after the path instead of hardcoding it.
9695 Find the stream based on the control string and only open a session when all
9698 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9700 * gst/rtsp-server/rtsp-media.c:
9701 * gst/rtsp-server/rtsp-media.h:
9702 media: add method to find a stream by control url
9704 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9706 * gst/rtsp-server/rtsp-stream.c:
9707 * gst/rtsp-server/rtsp-stream.h:
9708 stream: add method to check control url of stream
9710 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9712 * gst/rtsp-server/rtsp-client.c:
9713 * gst/rtsp-server/rtsp-session-media.c:
9714 * gst/rtsp-server/rtsp-session-media.h:
9715 * gst/rtsp-server/rtsp-session.c:
9716 * gst/rtsp-server/rtsp-session.h:
9717 session: use path matching for session media
9718 Use a path string instead of a uri to lookup session media in the sessions. Also
9719 use path matching to find the largest possible path that matches.
9721 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9723 * gst/rtsp-server/rtsp-client.c:
9724 * gst/rtsp-server/rtsp-mount-points.c:
9725 * gst/rtsp-server/rtsp-mount-points.h:
9726 * tests/check/gst/mountpoints.c:
9727 mount-points: remove useless vmethod
9728 Making lookups in the mount points should not be done with a URL, if there is a
9729 mapping to be done from URL to mount points, we'll need to do it somewhere
9732 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9734 * gst/rtsp-server/rtsp-mount-points.c:
9735 * gst/rtsp-server/rtsp-mount-points.h:
9736 * tests/check/gst/mountpoints.c:
9737 mount-points: improve mount point searching
9738 Use a GSequence to keep track of the mount points.
9739 Match a URL to the longest matching registered mount point. This should be the
9740 URL to perform aggreagate control and the remainder is the stream specific
9742 Add some unit tests for this.
9744 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9746 * gst/rtsp-server/Makefile.am:
9747 rtsp-server: Allow building of static library
9749 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9751 * tests/check/gst/mediafactory.c:
9752 tests: fix compilation
9754 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9756 * gst/rtsp-server/rtsp-sdp.c:
9757 sdp: get control string from stream
9758 Use the control string as configured in the stream.
9760 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9762 * gst/rtsp-server/rtsp-stream.c:
9763 * gst/rtsp-server/rtsp-stream.h:
9764 stream: add methods and property to set control string
9766 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9768 * gst/rtsp-server/rtsp-client.c:
9770 Rename variables for clarity
9771 Keep media in state when we can
9773 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9775 * gst/rtsp-server/rtsp-client.c:
9776 * gst/rtsp-server/rtsp-stream.c:
9777 * gst/rtsp-server/rtsp-stream.h:
9778 stream: add more support for IPv6
9779 Rename _get_address to _get_multicast_address in GstRTSPStream to
9780 make it clear that this function only deals with multicast.
9781 Make it possible to have both an IPv4 and IPv6 multicast address on
9782 a stream. Give the client an IPv4 or IPv6 address depending on the
9783 address it used to connect to the server.
9784 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9786 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9788 * gst/rtsp-server/rtsp-client.c:
9791 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9793 * gst/rtsp-server/rtsp-stream.c:
9794 stream: handle failed port allocation
9795 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9796 can't allocate any family at all. Also keep track of what port families we
9798 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9800 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9802 * gst/rtsp-server/rtsp-stream.c:
9803 stream: improve docs
9805 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9807 * gst/rtsp-server/rtsp-stream-transport.c:
9808 stream-transport: remove old if 0 block
9810 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9812 * tests/check/gst/client.c:
9814 gst_rtsp_client_get_uri() has been removed
9815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9817 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9819 * gst/rtsp-server/rtsp-client.c:
9820 * gst/rtsp-server/rtsp-client.h:
9821 client: add method to filter managed sessions
9822 Add a method to filter the sessions managed by this client connection.
9823 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9825 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9827 * gst/rtsp-server/rtsp-client.c:
9828 * gst/rtsp-server/rtsp-client.h:
9829 client: remove _get_uri() method
9830 Remove the get_uri() method on the client. A client has no uri, the uri
9831 property is an internal property to manage the last cached media for
9834 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9836 * gst/rtsp-server/rtsp-media-factory.h:
9837 media-factory: fix typo
9839 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9841 * gst/rtsp-server/rtsp-media.c:
9842 rtsp-media: Do not leak the query in default_query_stop
9843 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9845 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9847 * gst/rtsp-server/rtsp-media.c:
9848 media: don't unlock when conversion fails
9849 Don't unlock the state lock when conversion fails because it was not locked.
9851 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9853 * gst/rtsp-server/rtsp-media.c:
9854 * gst/rtsp-server/rtsp-media.h:
9855 Add query_position and query_stop vmethods to rtsp-media
9857 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9859 * gst/rtsp-server/rtsp-media.c:
9860 Fix typo in property install for rtsp-media's time-provider
9862 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9864 * gst/rtsp-server/rtsp-client.c:
9865 * gst/rtsp-server/rtsp-client.h:
9866 client: clean some variables
9867 Clean some variables and add some guards to _send_request()
9869 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9871 * gst/rtsp-server/rtsp-client.c:
9872 * gst/rtsp-server/rtsp-client.h:
9873 Add gst_rtsp_client_send_request API
9874 This makes it possible to send arbitrary messages to a client, such as
9875 SET_PARAMETER or GET_PARAMETER
9877 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9879 * gst/rtsp-server/rtsp-media.c:
9880 * gst/rtsp-server/rtsp-media.h:
9881 media: add _get_element() method
9882 Add method to get the element used when creating the media.
9883 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9885 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9887 * gst/rtsp-server/rtsp-media.c:
9890 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9892 * gst/rtsp-server/rtsp-stream.c:
9893 * gst/rtsp-server/rtsp-stream.h:
9894 stream: allow access to the rtp session
9895 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9897 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9899 * gst/rtsp-server/rtsp-stream.c:
9900 * gst/rtsp-server/rtsp-stream.h:
9901 dscp qos support in gst-rtsp-stream
9902 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9904 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9906 * tests/check/gst/rtspserver.c:
9908 Actually do what the comment says. Also keep the old code around, not sure what
9909 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9910 it currently doesn't.
9912 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9914 * gst/rtsp-server/rtsp-client.c:
9915 client: also watch newly created session
9916 When we newly created a session, start watching it immediately instead of
9917 on the next request.
9919 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9921 * tests/check/gst/client.c:
9922 tests: add unit test for new-session
9923 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9925 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9927 * gst/rtsp-server/rtsp-client.c:
9928 client: emit new-session when new session is created
9929 Only emit new-session when we created a new session for a client, not when a
9930 client picked up a previous session.
9931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9933 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9935 * gst/rtsp-server/rtsp-client.c:
9936 client: handle asterisk as path in requests
9937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9939 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9941 * gst/rtsp-server/rtsp-media.c:
9942 media: handle segment query format mismatch
9943 It's possible that the segment query returns with a different format than what
9944 we asked for, handle this case also.
9946 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9948 * gst/rtsp-server/rtsp-media.c:
9949 media: use segment stop in collect_media_stats
9950 Use segment stop instead of duration as range end point.
9951 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9953 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9955 * gst/rtsp-server/rtsp-media.c:
9956 * tests/check/gst/media.c:
9957 rtsp-media: Do not leak the element in take_pipeline
9958 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9960 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9962 * gst/rtsp-server/rtsp-client.c:
9963 * gst/rtsp-server/rtsp-client.h:
9964 rtsp-client: Make configure_client_transport virtual
9965 This patch makes configure_client_transport virtual. The functionality is
9966 needed to handle some weird clients sending multicast transport settings as url
9968 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9970 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9972 * gst/rtsp-server/rtsp-client.c:
9973 * gst/rtsp-server/rtsp-client.h:
9974 rtsp-client: Make param_set and param_get virtual
9975 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9977 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9979 * gst/rtsp-server/rtsp-client.c:
9980 * gst/rtsp-server/rtsp-media.c:
9981 * gst/rtsp-server/rtsp-media.h:
9982 media: convert_range replaces get_range_times
9983 get_range_times worked for handling UTC ranges for seeks, but we also
9984 need to convert back from NPT to the requested unit in
9985 get_range_string. convert_range is now used for both.
9986 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9988 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9990 * gst/rtsp-server/rtsp-client.c:
9991 * gst/rtsp-server/rtsp-sdp.c:
9992 * gst/rtsp-server/rtsp-sdp.h:
9993 sdp: cleanup sdp info
9994 We don't need to pass the proto, we can more easily check a boolean.
9995 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9997 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9999 * gst/rtsp-server/rtsp-sdp.c:
10000 use 0.0.0.0 or :: for c= line instead of server address
10002 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
10004 * gst/rtsp-server/rtsp-client.c:
10005 use local address, not remote, in SDP
10006 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
10008 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10011 Automatic update of common submodule
10012 From 098c0d7 to 01a7a46
10014 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
10016 * gst/rtsp-server/rtsp-media.c:
10017 * gst/rtsp-server/rtsp-media.h:
10018 media: possibility to override range time conversion
10019 Make it possible to override the conversion from GstRTSPTimeRange to
10020 GstClockTimes, that is done before seeking on the media
10021 pipeline. Overriding can be useful for UTC ranges, where the default
10022 conversion gives nanoseconds since 1900.
10023 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
10025 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
10027 * gst/rtsp-server/rtsp-server.c:
10028 * gst/rtsp-server/rtsp-server.h:
10029 rtsp-server: Expose the use_client_settings API
10030 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
10032 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
10034 * gst/rtsp-server/rtsp-client.c:
10035 * gst/rtsp-server/rtsp-stream.c:
10036 * gst/rtsp-server/rtsp-stream.h:
10037 rtspstream: handle both ipv4 and ipv6 clients
10038 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
10040 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10042 * gst/rtsp-server/rtsp-sdp.c:
10043 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
10044 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
10045 We already have a way to place extra attributes in the SDP by using a string
10046 property with prefix x- or a- in the caps.
10048 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10050 * gst/rtsp-server/rtsp-sdp.c:
10051 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
10052 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
10053 We already have a way to place extra attributes in the SDP, just make a string
10054 property in the payloader with a- or x- prefix.
10056 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10058 * gst/rtsp-server/rtsp-sdp.c:
10059 rtsp: place a- and x- properties as attributes
10060 application/x-rtp has properties with a- and x- prefixes that should be
10061 placed as attributes in the SDP for the media instead of being added to the
10064 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10066 * examples/Makefile.am:
10067 * examples/test-video.c:
10068 example: add TLS example
10070 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10072 * gst/rtsp-server/rtsp-server.c:
10073 * gst/rtsp-server/rtsp-server.h:
10074 server: add support for TLS
10075 Add methods to set and get a TLS certificate.
10076 Add vmethod to configure a new connection. By default, configure the TLS
10077 certificate in a new connection if needed.
10079 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10081 * gst/rtsp-server/rtsp-server.c:
10082 * gst/rtsp-server/rtsp-server.h:
10083 server: remove accept_client vmethod
10084 This vmethod is not very useful so remove it.
10086 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10088 * gst/rtsp-server/rtsp-server.c:
10089 server: don't crash on NULL GError
10091 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
10093 * gst/rtsp-server/rtsp-session-pool.c:
10094 rtsp-session-pool: corrected session timeout detection
10095 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
10097 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10099 * gst/rtsp-server/rtsp-client.c:
10100 client: improve debug
10102 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10104 * gst/rtsp-server/rtsp-client.c:
10105 * gst/rtsp-server/rtsp-client.h:
10106 * gst/rtsp-server/rtsp-server.c:
10107 server: refactor connection setup
10108 Let the server accept the socket connection and construct a GstRTSPConnection
10109 from it. Remove the code from the client and let the client only deal with
10110 a fully configure GstRTSPConnection object.
10111 We will need this later when the server will configure the connection for
10114 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10116 * gst/rtsp-server/rtsp-stream.c:
10117 stream: keep the transport object alive
10118 Keep the transport object alive while we have it as qdata on the
10121 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
10123 * gst/rtsp-server/rtsp-client.c:
10124 * gst/rtsp-server/rtsp-server.c:
10125 rtsp-server: Do not crash on nmapping of server
10126 * generate error when gst_rtsp_connection_accept fails
10127 * do not stop accepting incoming connections because
10128 accepting a client fails
10129 https://bugzilla.gnome.org/show_bug.cgi?id=701072
10131 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
10133 * gst/rtsp-server/rtsp-client.c:
10134 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
10135 https://bugzilla.gnome.org/show_bug.cgi?id=700953
10137 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
10139 * gst/rtsp-server/rtsp-sdp.c:
10140 rtsp-sdp: Parse framerate caps field and set SDP attribute
10141 The SDP attribute and its format is described in RFC4566.
10142 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10144 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
10146 * gst/rtsp-server/rtsp-sdp.c:
10147 rtsp-sdp: Parse width/height from caps and set SDP attribute
10148 The SDP attribute and its format is described in RFC6064.
10149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
10151 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
10153 * gst/rtsp-server/rtsp-sdp.c:
10154 * tests/check/gst/client.c:
10155 rtsp-sdp: add bandwidth line
10156 https://bugzilla.gnome.org/show_bug.cgi?id=699220
10158 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10161 Automatic update of common submodule
10162 From 5edcd85 to 098c0d7
10164 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10166 * tests/check/gst/media.c:
10167 tests: add dynamic payloader prepare/unprepare check
10169 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10171 * gst/rtsp-server/rtsp-media.c:
10172 media: release lock when removing fakesink
10174 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10176 * gst/rtsp-server/rtsp-stream.c:
10177 stream: set elements to NULL before removing
10178 When removing a stream, set the elements to NULL first. This avoids
10179 element-is-not-in-NULL-state errors when we dispose the elements.
10181 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
10184 Automatic update of common submodule
10185 From 3cb3d3c to 5edcd85
10187 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10189 * gst/rtsp-server/rtsp-media.c:
10190 * gst/rtsp-server/rtsp-media.h:
10191 media: listen to pad-removed signals
10192 Listen to the pad-removed signal and remove the stream associated with the
10194 Add signal to be notified of the removed pad.
10195 Remove the fakesink in unprepare()
10196 Fix signatures of the signal methods
10198 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10200 * examples/test-sdp.c:
10201 tests: add example of reusable pipelines
10203 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
10205 * gst/rtsp-server/rtsp-stream.c:
10206 * gst/rtsp-server/rtsp-stream.h:
10207 stream: add method to get the srcpad
10209 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
10211 * tests/check/gst/media.c:
10212 check: add media prepare/unprepare test
10213 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10215 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
10217 * gst/rtsp-server/rtsp-media.c:
10218 media: disconnect from signal handlers in unprepare()
10219 We connected to the pad-added and no-more-pads signals in prepare() so
10220 we need to disconnect from them in unprepare().
10221 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10223 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
10225 * gst/rtsp-server/rtsp-media.c:
10226 media: don't free streams array
10227 Don't free the streams array in the unprepare() method, they were not
10228 added in prepare().
10229 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10231 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
10233 * gst/rtsp-server/rtsp-media.c:
10234 media: don't unref the pipeline in unprepare
10235 Unprepare() should undo what prepare() does. Because the pipeline is
10236 not created in prepare(), we should not unref it in unprepare()
10238 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
10240 * gst/rtsp-server/rtsp-stream.c:
10241 stream: clear session and caps for reuse
10242 Set the session and caps to NULL after unref otherwise we might unref
10244 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
10246 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
10248 * gst/rtsp-server/rtsp-client.c:
10249 client: send out teardown signal before tearing down
10250 The advantage is that in the signal handler you get direct access to
10251 information about what streams are about to get torn down (in the
10252 GstRTSPClientState).
10253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
10255 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
10257 * gst/rtsp-server/rtsp-client.c:
10258 * gst/rtsp-server/rtsp-client.h:
10259 client: expose connection
10260 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
10262 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
10265 Automatic update of common submodule
10266 From aed87ae to 3cb3d3c
10268 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10270 * gst/rtsp-server/rtsp-media.c:
10271 * gst/rtsp-server/rtsp-media.h:
10272 * gst/rtsp-server/rtsp-session-media.c:
10273 * gst/rtsp-server/rtsp-session-media.h:
10274 media: add method to get the base_time of the pipeline
10275 Together with a shared clock, this base-time could eventually be sent to
10276 the client so that it can reconstruct the exact running-time of the clock
10279 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10281 * gst/rtsp-server/Makefile.am:
10282 * gst/rtsp-server/rtsp-media.c:
10283 * gst/rtsp-server/rtsp-media.h:
10284 * gst/rtsp-server/rtsp-sdp.c:
10285 media: add GstNetTimeProvider support
10286 Add a property to let the media provide a GstNetTimeProvider for its clock.
10287 Make methods to get the clock and nettimeprovider
10288 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
10289 provider and also the current time of the clock. This should make it possible
10290 for (GStreamer) clients to slave their clock to the server clock.
10292 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
10295 Automatic update of common submodule
10296 From 04c7a1e to aed87ae
10298 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10300 * gst/rtsp-server/rtsp-media.c:
10301 media: wait for buffering to complete
10302 Wait for buffering to complete before changing the state to the target state.
10304 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10306 * gst/rtsp-server/rtsp-media.c:
10307 media: small cleanup
10309 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
10311 * tests/check/gst/rtspserver.c:
10312 tests: remove extra unref in test_setup_non_existing_stream
10313 The unref is not needed anymore, teardown runs without it.
10314 https://bugzilla.gnome.org/show_bug.cgi?id=696542
10316 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
10318 * tests/check/gst/rtspserver.c:
10319 tests: GSocketService cleanup in test_bind_already_in_use
10320 Use g_socket_service_stop so the rtspserver test stops listening for
10321 incoming connections in test_bind_already_in_use.
10322 https://bugzilla.gnome.org/show_bug.cgi?id=696541
10324 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
10326 * gst/rtsp-server/rtsp-media-factory.c:
10327 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
10328 Instead use a GWeakRef which is safe to use
10329 This is a known GLib bug, see:
10330 https://bugzilla.gnome.org/show_bug.cgi?id=667145
10332 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
10334 * gst/rtsp-server/rtsp-client.c:
10335 * gst/rtsp-server/rtsp-media.c:
10336 * gst/rtsp-server/rtsp-media.h:
10337 * gst/rtsp-server/rtsp-sdp.c:
10338 * tests/check/gst/media.c:
10339 * tests/check/gst/rtspserver.c:
10340 rtsp-media/client: Reply to PLAY request with same type of Range
10341 Remember the type of Range from the PLAY request and use the same type for
10344 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
10346 * gst/rtsp-server/rtsp-client.c:
10347 * gst/rtsp-server/rtsp-client.h:
10348 * tests/check/gst/client.c:
10349 rtsp-client: expose uri
10351 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
10353 * tests/check/gst/mediafactory.c:
10354 tests: Hold ref while creating second media
10355 To test if the media aren't shared, make sure we keep the first one while creating a second
10356 otherwise the same memory address may be reused.
10358 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
10361 configure: remove out-of-date comment
10363 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
10366 .gitignore: ignore more build files
10368 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10370 * tests/check/Makefile.am:
10371 tests: use right _LIBS variable for gst-plugins-base libs
10373 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10375 * tests/check/Makefile.am:
10376 check: add librtp to libs
10378 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
10380 * tests/check/gst/rtspserver.c:
10381 tests: Add test to check selecting a port the server will send from
10383 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
10385 * tests/check/gst/rtspserver.c:
10386 tests: Make sure packets are actually received
10388 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10390 * gst/rtsp-server/rtsp-stream.c:
10391 stream: Select unicast address from pool if appropriate
10393 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
10395 * gst/rtsp-server/rtsp-stream.c:
10396 stream: Properties are always there in Gst 1.0
10398 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10400 * tests/check/gst/addresspool.c:
10401 tests: Add tests for unicast addresses in pool
10403 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
10405 * gst/rtsp-server/rtsp-address-pool.c:
10406 * tests/check/gst/addresspool.c:
10407 address-pool: Verify that multicast addresses are used for multicast and vice-versa
10409 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
10411 * docs/libs/gst-rtsp-server-sections.txt:
10412 * gst/rtsp-server/rtsp-address-pool.c:
10413 * gst/rtsp-server/rtsp-address-pool.h:
10414 * gst/rtsp-server/rtsp-stream.c:
10415 * tests/check/gst/addresspool.c:
10416 address-pool: Add unicast addresses
10418 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10421 * gst/rtsp-server/rtsp-server.c:
10422 * tests/check/gst/rtspserver.c:
10423 rtsp-server: Limit the number of threads per server instance
10424 If we exceed the maximum, just round robin the clients over the existing
10427 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
10429 * gst/rtsp-server/rtsp-server.c:
10430 rtsp-server: No need to store the GMainContext in the client context
10432 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
10434 * tests/check/gst/rtspserver.c:
10435 tests: Add test for client disconnection
10437 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
10439 * tests/check/gst/rtspserver.c:
10440 tests: Test client and session timeouts with multiple threads
10442 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
10444 * gst/rtsp-server/rtsp-address-pool.c:
10445 * gst/rtsp-server/rtsp-auth.c:
10446 * gst/rtsp-server/rtsp-client.c:
10447 * gst/rtsp-server/rtsp-media-factory-uri.c:
10448 * gst/rtsp-server/rtsp-media-factory.c:
10449 * gst/rtsp-server/rtsp-media.c:
10450 * gst/rtsp-server/rtsp-mount-points.c:
10451 * gst/rtsp-server/rtsp-server.c:
10452 * gst/rtsp-server/rtsp-session-media.c:
10453 * gst/rtsp-server/rtsp-session-pool.c:
10454 * gst/rtsp-server/rtsp-session.c:
10455 Document locking and its order
10457 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
10459 * tests/check/gst/rtspserver.c:
10460 tests: Test that slow DESCRIBE don't block other clients
10462 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
10464 * tests/check/gst/client.c:
10465 tests: Add tests for client-requested multicast address
10467 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
10469 * docs/libs/gst-rtsp-server-sections.txt:
10470 docs: Put the various functions in the right sections
10472 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
10474 * docs/libs/gst-rtsp-server-docs.sgml:
10475 * docs/libs/gst-rtsp-server-sections.txt:
10476 * gst/rtsp-server/rtsp-address-pool.c:
10477 * gst/rtsp-server/rtsp-address-pool.h:
10478 docs: Generate docs for GstRTSPAddressPool
10480 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
10482 * gst/rtsp-server/rtsp-client.c:
10483 * gst/rtsp-server/rtsp-stream.c:
10484 * gst/rtsp-server/rtsp-stream.h:
10485 client: Check client provided addresses against the address pool
10487 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
10489 * gst/rtsp-server/rtsp-address-pool.c:
10490 * gst/rtsp-server/rtsp-address-pool.h:
10491 * tests/check/gst/addresspool.c:
10492 address-pool: Add API to request a specific address from the pool
10493 Also add relevant unit tests.
10495 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
10497 * tests/check/gst/mediafactory.c:
10498 tests: Check the passing around of a RTSPAddressPool
10499 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
10500 way down to the stream.
10502 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
10504 * tests/check/gst/addresspool.c:
10505 tests: Add more tests for the address pool
10507 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
10509 * gst/rtsp-server/rtsp-address-pool.c:
10510 address-pool: Fix off by one error
10511 When splitting a port range, the port after a skip is not part of range.
10513 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
10516 Automatic update of common submodule
10517 From 2de221c to 04c7a1e
10519 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
10522 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
10523 AM_CONFIG_HEADER was removed in automake 1.13
10524 https://bugzilla.gnome.org/show_bug.cgi?id=693368
10526 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
10529 Automatic update of common submodule
10530 From a942293 to 2de221c
10532 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10534 * gst/rtsp-server/rtsp-client.c:
10535 client: make sure the watch exists while sending data
10536 Protect the send_func with a lock. This allows us to wait for sending
10537 to complete before changing the send_func and user_data. We add an
10538 extra ref to the watch to make sure that it remains valid during
10540 When closing the connection, set the send_func to NULL
10541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
10543 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10545 * tests/check/Makefile.am:
10546 tests: use GST_*_1_0 environment variables everywhere
10547 The _1_0 suffixed environment variables override the
10548 non-suffixed ones, so if we're in an environment that
10549 sets the _1_0 suffixed ones, such as jhbuild, we need
10550 to set those to make sure ours actually always get
10553 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10556 Automatic update of common submodule
10557 From acb04d9 to a942293
10559 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10561 * gst/rtsp-server/rtsp-client.c:
10562 rtsp-client: set the client backlog
10563 Set the client backlog to a reasonable default
10565 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
10567 * gst/rtsp-server/rtsp-media.c:
10568 rtsp-media: Make the element a constructor parameter
10569 https://bugzilla.gnome.org/show_bug.cgi?id=689594
10571 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
10573 * docs/libs/Makefile.am:
10574 docs: Link with gcov library when gcov is enabled
10575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
10577 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10579 * gst/rtsp-server/rtsp-media.c:
10580 media: match prepare with unprepare
10581 Really unprepare when there were an equal amount of prepare calls.
10583 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10585 * gst/rtsp-server/rtsp-media.c:
10586 media: media has to be unprepared in finalize
10587 Because unprepare takes away the last ref on the media.
10589 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10591 * gst/rtsp-server/rtsp-client.c:
10592 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
10593 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
10594 We can't use the refcount to trigger unprepare because it is the unprepare call
10595 that removes the last refcount after all messages are consumed. What we should
10596 probably do is make a prepared refcount and only unprepare when the refcount
10599 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10601 * gst/rtsp-server/rtsp-media.c:
10602 media: let the source unref the last media ref
10603 the last ref to the media is held by the source so we don't need to add more ref
10604 and unrefs, we simply destroy the media when the source is gone.
10606 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10608 * gst/rtsp-server/rtsp-media.c:
10609 media: improve debug
10611 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10613 * gst/rtsp-server/rtsp-media.c:
10615 Make sure we are in the right state when collecting the position and duration.
10616 Only make ourselves PREPARED when we were previously PREPARING.
10618 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10620 * gst/rtsp-server/rtsp-media.c:
10621 media: use g_object_ref/unref for GObjects
10623 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10625 * gst/rtsp-server/rtsp-client.c:
10626 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10627 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10628 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10629 isn't being used anymore.
10631 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10633 * gst/rtsp-server/rtsp-media.c:
10634 Fix compiler warning
10636 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10638 * gst/rtsp-server/rtsp-media-factory-uri.c:
10639 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10641 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10643 * gst/rtsp-server/rtsp-session-media.h:
10646 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10648 * gst/rtsp-server/rtsp-media.c:
10649 * tests/check/gst/media.c:
10650 media: avoid element leak
10652 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10654 * gst/rtsp-server/rtsp-media.c:
10655 media: require an element in media constructor
10657 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10659 * gst/rtsp-server/rtsp-client.c:
10660 Revert "client: TEARDOWN brings that state to Init again"
10661 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10662 The object is already disposed, there is no point in setting the state.
10664 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10666 * gst/rtsp-server/rtsp-client.c:
10667 client: TEARDOWN brings that state to Init again
10669 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10671 * docs/libs/gst-rtsp-server-sections.txt:
10672 * examples/test-auth.c:
10673 * gst/rtsp-server/rtsp-auth.c:
10674 * gst/rtsp-server/rtsp-auth.h:
10675 * gst/rtsp-server/rtsp-client.c:
10676 * gst/rtsp-server/rtsp-client.h:
10677 * gst/rtsp-server/rtsp-media-factory-uri.c:
10678 * gst/rtsp-server/rtsp-media-factory-uri.h:
10679 * gst/rtsp-server/rtsp-media-factory.c:
10680 * gst/rtsp-server/rtsp-media-factory.h:
10681 * gst/rtsp-server/rtsp-media.c:
10682 * gst/rtsp-server/rtsp-media.h:
10683 * gst/rtsp-server/rtsp-mount-points.c:
10684 * gst/rtsp-server/rtsp-mount-points.h:
10685 * gst/rtsp-server/rtsp-sdp.c:
10686 * gst/rtsp-server/rtsp-server.c:
10687 * gst/rtsp-server/rtsp-server.h:
10688 * gst/rtsp-server/rtsp-session-media.c:
10689 * gst/rtsp-server/rtsp-session-media.h:
10690 * gst/rtsp-server/rtsp-session-pool.c:
10691 * gst/rtsp-server/rtsp-session-pool.h:
10692 * gst/rtsp-server/rtsp-session.c:
10693 * gst/rtsp-server/rtsp-session.h:
10694 * gst/rtsp-server/rtsp-stream-transport.c:
10695 * gst/rtsp-server/rtsp-stream-transport.h:
10696 * gst/rtsp-server/rtsp-stream.c:
10697 * gst/rtsp-server/rtsp-stream.h:
10698 * tests/check/gst/media.c:
10699 rtsp: make object details private
10700 Make all object details private
10701 Add methods to access private bits
10703 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10705 * tests/check/Makefile.am:
10706 * tests/check/gst/media.c:
10707 tests: add media tests
10709 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10711 * gst/rtsp-server/rtsp-media.c:
10712 media: check if prepared for some methods
10713 Check that the media object is prepared before doing seek and getting the
10714 current position etc.
10715 Add some g_return checks.
10717 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10719 * tests/check/Makefile.am:
10720 * tests/check/gst/mediafactory.c:
10721 tests: add mediafactory test
10723 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10725 * gst/rtsp-server/rtsp-stream.c:
10726 stream: improve debug
10728 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10730 * gst/rtsp-server/rtsp-media.c:
10731 * gst/rtsp-server/rtsp-media.h:
10732 media: unref pipeline in finalize to avoid leaking it
10734 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10736 * gst/rtsp-server/rtsp-media-factory-uri.c:
10737 * gst/rtsp-server/rtsp-media.c:
10738 rtsp: use gst_object_unref on GstObjects
10740 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10742 * gst/rtsp-server/rtsp-media-factory.c:
10743 media-factory: require an url
10745 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10747 * examples/test-uri.c:
10748 examples: fix include
10750 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10752 * gst/rtsp-server/rtsp-server.h:
10753 server: remove unused include
10755 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10757 * tests/check/Makefile.am:
10758 * tests/check/gst/mountpoints.c:
10759 tests: add test for mountpoints
10761 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10763 * gst/rtsp-server/rtsp-client.c:
10764 client: fix factory leak
10765 Keep the factory in the state object only for authorization checks and make
10766 sure we unref it on failure. Also don't keep invalid objects in the state
10769 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10771 * gst/rtsp-server/rtsp-mount-points.c:
10772 mounts: add g_return_if guards
10774 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10776 * tests/check/gst/client.c:
10777 tests: add more tests
10779 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10781 * gst/rtsp-server/rtsp-client.c:
10782 client: improve debug
10784 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10786 * gst/rtsp-server/rtsp-client.c:
10787 client: improve debug and fix leaks
10788 Cleanup the uri and session when there is a bad request.
10790 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10795 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10797 * tests/check/gst/client.c:
10798 test: add test for session in options request
10800 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-client.c:
10803 client: use 454 when session can't be found
10804 We should use 454 when a session can't be found because there was no session
10805 pool configured in the server. This is not a server configuration problem
10806 because the server on which the request is done might not be the same one that
10807 will keep the sessions for us and so it does not need to support sessions.
10809 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10811 * gst/rtsp-server/rtsp-client.c:
10812 client: only free connection when there is one
10813 It's possible that the client doesn't have a connection when we try to free it.
10815 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10817 * tests/check/Makefile.am:
10818 * tests/check/gst/client.c:
10819 tests: add unit test for the client object
10821 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10823 * gst/rtsp-server/rtsp-client.c:
10824 client: small cleanup
10826 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10828 * gst/rtsp-server/rtsp-client.h:
10829 client: remove unused include
10831 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10833 * gst/rtsp-server/rtsp-client.c:
10834 client: fix compilation
10836 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10838 * gst/rtsp-server/rtsp-client.c:
10839 client: call destroy without the lock
10841 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10843 * gst/rtsp-server/rtsp-client.c:
10844 * gst/rtsp-server/rtsp-client.h:
10845 client: make the client usable without a socket
10846 Make a method to let the client handle a message and a callback when the client
10847 wants us to send a response message back. This makes it possible to also use the
10848 client object without the sockets, which should make it easier to test.
10850 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10852 * gst/rtsp-server/rtsp-client.c:
10853 * gst/rtsp-server/rtsp-client.h:
10854 client: small cleanup
10856 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10858 * docs/libs/gst-rtsp-server-sections.txt:
10859 * gst/rtsp-server/rtsp-client.c:
10860 * gst/rtsp-server/rtsp-client.h:
10861 * gst/rtsp-server/rtsp-server.c:
10862 client: remove reference to server
10863 We don't need to keep a ref to the server
10865 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10867 * gst/rtsp-server/rtsp-client.c:
10868 * gst/rtsp-server/rtsp-client.h:
10869 client: add locking
10870 Also add some g_return_if()
10872 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10874 * gst/rtsp-server/rtsp-client.c:
10875 client: log more errors
10877 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10879 * gst/rtsp-server/rtsp-client.c:
10880 client: fix compilation
10882 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10884 * gst/rtsp-server/rtsp-client.c:
10885 * gst/rtsp-server/rtsp-client.h:
10886 client: add generic close-after-send support
10887 Add a property to send_response() to close the connection after the response has
10888 been sent to the client.
10890 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10893 * docs/libs/gst-rtsp-server-docs.sgml:
10894 * docs/libs/gst-rtsp-server-sections.txt:
10895 * docs/libs/gst-rtsp-server.types:
10896 * examples/test-auth.c:
10897 * examples/test-launch.c:
10898 * examples/test-mp4.c:
10899 * examples/test-multicast.c:
10900 * examples/test-multicast2.c:
10901 * examples/test-ogg.c:
10902 * examples/test-readme.c:
10903 * examples/test-sdp.c:
10904 * examples/test-uri.c:
10905 * examples/test-video.c:
10906 * gst/rtsp-server/Makefile.am:
10907 * gst/rtsp-server/rtsp-auth.h:
10908 * gst/rtsp-server/rtsp-client.c:
10909 * gst/rtsp-server/rtsp-client.h:
10910 * gst/rtsp-server/rtsp-media-mapping.c:
10911 * gst/rtsp-server/rtsp-media-mapping.h:
10912 * gst/rtsp-server/rtsp-mount-points.c:
10913 * gst/rtsp-server/rtsp-mount-points.h:
10914 * gst/rtsp-server/rtsp-server.c:
10915 * gst/rtsp-server/rtsp-server.h:
10916 * gst/rtsp-server/rtsp-session-media.c:
10917 * gst/rtsp-server/rtsp-session-pool.c:
10918 * gst/rtsp-server/rtsp-session-pool.h:
10919 * tests/check/gst/rtspserver.c:
10920 MediaMapping -> MountPoints
10921 Describes better what the object manages.
10923 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10926 configure: bump required version of -base
10928 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10930 * gst/rtsp-server/rtsp-media.c:
10933 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10935 * gst/rtsp-server/rtsp-media.c:
10936 * gst/rtsp-server/rtsp-media.h:
10937 media: support more Range formats
10938 Use the new -base methods to convert the Range string into a seek start and stop
10941 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10943 * examples/test-launch.c:
10944 examples: fix whitespace
10946 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10948 * examples/test-auth.c:
10949 test-auth: add example of how to remove sessions
10950 Add an example of the session filter api.
10952 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10954 * examples/test-uri.c:
10955 test-uri: remove mapping example
10957 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10959 * examples/test-uri.c:
10960 test-uri: fix callback signature
10962 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10964 * gst/rtsp-server/rtsp-media-factory.c:
10965 factory: keep ref to factory while media active
10966 While the media from a factory is alive, keep a ref to the factory.
10967 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10969 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10971 * gst/rtsp-server/rtsp-media-factory-uri.c:
10972 factory-uri: add some debug
10974 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10976 * gst/rtsp-server/rtsp-stream.c:
10977 stream: set udp sources to PLAYING
10978 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10979 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10981 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10983 * gst/rtsp-server/rtsp-media-factory-uri.c:
10984 factory-uri: take ref to factory
10985 Take a ref to the factory that we place in our list.
10987 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10989 * tests/Makefile.am:
10990 * tests/test-reuse.c:
10991 test: add test for server reuse
10992 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10994 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10996 * gst/rtsp-server/rtsp-server.c:
10997 server: start and stop multiple times
10998 Stop listening on the RTSP port when the GSource is removed, so clients
10999 can't connect and the server can be started again.
11000 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
11002 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11004 * gst/rtsp-server/rtsp-server.c:
11005 server: fix small leak
11007 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11009 * gst/rtsp-server/rtsp-media.c:
11010 media: unref source in finish_unprepare
11011 The source is created in prepare, unref it in finish_unprepare.
11012 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
11014 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
11016 * gst/rtsp-server/rtsp-client.c:
11017 * gst/rtsp-server/rtsp-media.c:
11018 rtsp-media: remove bus watch before finalizing
11019 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
11020 * An extra media ref is added for the bus watch. This extra ref is unreffed by
11021 the GDestroyNotify function.
11022 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
11023 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
11024 gst_rtsp_media_unprepare before unreffing the media.
11025 This way, the bus watch will be removed before the media is finalized.
11026 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
11028 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
11030 * gst/rtsp-server/rtsp-client.c:
11031 * gst/rtsp-server/rtsp-client.h:
11032 client: wait until the TEARDOWN response is sent to close the connection
11033 Responses can be sent async so we need to wait until the TEARDOWN response has
11034 been written before we close the connection to the client. This avoids the risk
11035 of writing/polling closed sockets.
11036 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
11038 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
11040 * gst/rtsp-server/rtsp-stream.c:
11041 rtsp-stream: plug socket leak
11042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
11044 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
11047 Automatic update of common submodule
11048 From 6bb6951 to a72faea
11050 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
11052 * gst/rtsp-server/rtsp-media-factory-uri.c:
11053 rtsp-server: don't use deprecated API
11055 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
11057 * gst/rtsp-server/rtsp-client.c:
11058 rtsp-client: fix unused-but-set-variable compiler warning
11059 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
11061 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11064 * docs/libs/gst-rtsp-server-sections.txt:
11065 * gst/rtsp-server/rtsp-client.c:
11068 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11070 * examples/Makefile.am:
11071 * examples/test-multicast2.c:
11072 examples: add another multicast example
11073 Add an example for how to configure separate multicast ranges for each media
11076 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11078 * examples/test-multicast.c:
11081 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11083 * gst/rtsp-server/rtsp-client.c:
11084 * gst/rtsp-server/rtsp-media.c:
11085 * gst/rtsp-server/rtsp-session-media.c:
11086 * gst/rtsp-server/rtsp-session-media.h:
11087 * gst/rtsp-server/rtsp-stream-transport.c:
11088 * gst/rtsp-server/rtsp-stream-transport.h:
11089 stream: use the address managed by the stream
11090 Use the address managed by the stream for multicast. This allows us to have 1
11091 multicast address for each stream.
11092 Because the address is now managed by the stream we don't have to pass it around
11094 Set the address pool on the streams.
11096 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11098 * gst/rtsp-server/rtsp-client.c:
11099 * gst/rtsp-server/rtsp-media.c:
11100 * gst/rtsp-server/rtsp-stream.c:
11101 rtsp: improve debug
11103 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11105 * gst/rtsp-server/rtsp-media.c:
11106 * gst/rtsp-server/rtsp-media.h:
11107 media: add signal for new streams
11108 This allows applications to listen for new streams and configure properties on
11109 them, like the address pool.
11111 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11113 * gst/rtsp-server/rtsp-media.c:
11114 media: configure address pool in new streams
11116 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11118 * gst/rtsp-server/rtsp-stream.c:
11119 * gst/rtsp-server/rtsp-stream.h:
11120 stream: add methods to deal with address pool
11121 Add methods to get and set the address pool for the stream
11122 Add method to allocate and get the multicast addresses for this stream.
11124 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11126 * docs/libs/gst-rtsp-server-sections.txt:
11127 * gst/rtsp-server/rtsp-media.c:
11128 * gst/rtsp-server/rtsp-media.h:
11129 media: remove MTU property
11130 It is a stream property
11132 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11134 * gst/rtsp-server/rtsp-client.c:
11135 client: set blocksize only on stream
11136 Set the blocksize only on the current stream.
11138 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11140 * gst/rtsp-server/rtsp-stream.c:
11141 stream: share src and sink sockets
11142 the allocated socket is in the used-socket property, not socket.
11144 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11146 * gst/rtsp-server/rtsp-address-pool.c:
11147 * gst/rtsp-server/rtsp-address-pool.h:
11148 * gst/rtsp-server/rtsp-client.c:
11149 * gst/rtsp-server/rtsp-session-media.c:
11150 * gst/rtsp-server/rtsp-session-media.h:
11151 * gst/rtsp-server/rtsp-stream-transport.c:
11152 * gst/rtsp-server/rtsp-stream-transport.h:
11153 * tests/check/gst/addresspool.c:
11154 rtsp: make address-pool return an address object
11155 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
11156 store more info in the structure and allows us to more easily return the address
11157 to the right pool when no longer needed.
11158 Pass the address to the StreamTransport so that we can return it to the pool
11159 when the stream transport is freed or changed.
11161 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11163 * examples/Makefile.am:
11164 * examples/test-multicast.c:
11165 examples: add multicast example
11166 Show how to set up the multicast address pool so that media can be
11167 server with multicast.
11169 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11171 * gst/rtsp-server/rtsp-client.c:
11172 * gst/rtsp-server/rtsp-media-factory.c:
11173 * gst/rtsp-server/rtsp-media-factory.h:
11174 * gst/rtsp-server/rtsp-media.c:
11175 * gst/rtsp-server/rtsp-media.h:
11176 rtsp: use AddressPool
11177 Remove the multicast_group property.
11178 Use the configured addresspool to allocate multicast addresses.
11180 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11182 * gst/rtsp-server/rtsp-address-pool.c:
11183 * gst/rtsp-server/rtsp-address-pool.h:
11184 address-pool: add clear method
11186 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11188 * gst/rtsp-server/rtsp-address-pool.c:
11189 address-pool: small cleanups
11191 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11193 * tests/check/Makefile.am:
11194 * tests/check/gst/addresspool.c:
11195 tests: add addresspool unit test
11197 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11199 * gst/rtsp-server/Makefile.am:
11200 * gst/rtsp-server/rtsp-address-pool.c:
11201 * gst/rtsp-server/rtsp-address-pool.h:
11202 address-pool: add object to manage multicast addresses
11203 Make an object that can manage a rage of multicast addresses and ports.
11205 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11207 * gst/rtsp-server/rtsp-server.c:
11208 server: set default max-threads property
11210 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11212 * gst/rtsp-server/rtsp-media.c:
11213 media: wait for concurrent _prepare
11214 If a prepare is busy, wait for the result.
11216 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11218 * gst/rtsp-server/rtsp-media.c:
11219 media: add lock around message handler
11220 We don't want to dispatch messages while we are still processing the result of
11223 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11225 * gst/rtsp-server/rtsp-media.c:
11226 * gst/rtsp-server/rtsp-media.h:
11227 media: add lock to protect state changes
11229 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11231 * gst/rtsp-server/rtsp-stream.c:
11232 * gst/rtsp-server/rtsp-stream.h:
11233 stream: add locking
11235 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11237 * gst/rtsp-server/rtsp-stream-transport.c:
11238 * gst/rtsp-server/rtsp-stream-transport.h:
11239 * gst/rtsp-server/rtsp-stream.c:
11240 stream-transport: add keep-alive method
11242 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11244 * gst/rtsp-server/rtsp-stream-transport.c:
11245 * gst/rtsp-server/rtsp-stream-transport.h:
11246 * gst/rtsp-server/rtsp-stream.c:
11247 stream-transport: add method to handle RTP/RTCP
11248 Call new methods instead of poking into the structures directly.
11250 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11252 * gst/rtsp-server/rtsp-session-media.c:
11253 * gst/rtsp-server/rtsp-session-media.h:
11254 session-media: add locking
11256 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11258 * gst/rtsp-server/rtsp-session.c:
11259 * gst/rtsp-server/rtsp-session.h:
11260 session: add locking
11262 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11264 * gst/rtsp-server/rtsp-server.c:
11265 server: free old socket
11267 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11269 * gst/rtsp-server/rtsp-media-mapping.c:
11270 * gst/rtsp-server/rtsp-media-mapping.h:
11271 mapping: add locking
11273 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11275 * gst/rtsp-server/rtsp-media-factory.c:
11276 media-factory: add locking
11278 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11280 * gst/rtsp-server/rtsp-auth.c:
11281 * gst/rtsp-server/rtsp-auth.h:
11284 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11286 * gst/rtsp-server/rtsp-server.c:
11287 * gst/rtsp-server/rtsp-server.h:
11288 server: add max-thread property
11290 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11292 * gst/rtsp-server/rtsp-server.c:
11293 * gst/rtsp-server/rtsp-server.h:
11294 server: use a threadpool for the mainloops
11296 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11298 * gst/rtsp-server/rtsp-client.c:
11299 * gst/rtsp-server/rtsp-client.h:
11300 client: rename method
11301 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
11302 don't really create the client from the socket, we use the socket for the
11305 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11307 * gst/rtsp-server/rtsp-client.c:
11308 * gst/rtsp-server/rtsp-client.h:
11309 * gst/rtsp-server/rtsp-server.c:
11310 server: rework maincontext handling in clients
11311 Make a separate method to attach a client to a MainContext.
11312 Let the server decide in what GMainContext the client will operate and give this
11313 context to the client in attach. Then the server can later decide to use a
11314 separate thread for each client or just use the mainthread.
11316 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11318 * gst/rtsp-server/rtsp-client.c:
11319 * gst/rtsp-server/rtsp-session.c:
11320 * gst/rtsp-server/rtsp-session.h:
11321 session: move session header code in session object
11323 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
11327 * examples/test-auth.c:
11328 * examples/test-launch.c:
11329 * examples/test-mp4.c:
11330 * examples/test-ogg.c:
11331 * examples/test-readme.c:
11332 * examples/test-sdp.c:
11333 * examples/test-uri.c:
11334 * examples/test-video.c:
11335 * gst/rtsp-server/rtsp-auth.c:
11336 * gst/rtsp-server/rtsp-auth.h:
11337 * gst/rtsp-server/rtsp-client.c:
11338 * gst/rtsp-server/rtsp-client.h:
11339 * gst/rtsp-server/rtsp-media-factory-uri.c:
11340 * gst/rtsp-server/rtsp-media-factory-uri.h:
11341 * gst/rtsp-server/rtsp-media-factory.c:
11342 * gst/rtsp-server/rtsp-media-factory.h:
11343 * gst/rtsp-server/rtsp-media-mapping.c:
11344 * gst/rtsp-server/rtsp-media-mapping.h:
11345 * gst/rtsp-server/rtsp-media.c:
11346 * gst/rtsp-server/rtsp-media.h:
11347 * gst/rtsp-server/rtsp-params.c:
11348 * gst/rtsp-server/rtsp-params.h:
11349 * gst/rtsp-server/rtsp-sdp.c:
11350 * gst/rtsp-server/rtsp-sdp.h:
11351 * gst/rtsp-server/rtsp-server.c:
11352 * gst/rtsp-server/rtsp-server.h:
11353 * gst/rtsp-server/rtsp-session-media.c:
11354 * gst/rtsp-server/rtsp-session-media.h:
11355 * gst/rtsp-server/rtsp-session-pool.c:
11356 * gst/rtsp-server/rtsp-session-pool.h:
11357 * gst/rtsp-server/rtsp-session.c:
11358 * gst/rtsp-server/rtsp-session.h:
11359 * gst/rtsp-server/rtsp-stream-transport.c:
11360 * gst/rtsp-server/rtsp-stream-transport.h:
11361 * gst/rtsp-server/rtsp-stream.c:
11362 * gst/rtsp-server/rtsp-stream.h:
11363 * tests/check/gst/rtspserver.c:
11364 * tests/test-cleanup.c:
11367 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11369 * gst/rtsp-server/rtsp-media.c:
11370 * gst/rtsp-server/rtsp-session-media.c:
11371 * gst/rtsp-server/rtsp-session.c:
11372 rtsp-server: added annotations to indicate type of ownership transfer of return values
11373 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11375 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11378 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
11380 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
11383 * bindings/Makefile.am:
11384 * bindings/vala/Makefile.am:
11385 * bindings/vala/gst-rtsp-server-0.10.deps:
11386 * bindings/vala/gst-rtsp-server-0.10.vapi:
11387 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11388 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11389 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11390 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11391 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11393 bindings: remove vala bindings
11394 They'll be reunited with the other GStreamer bindings
11395 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11397 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11399 * gst/rtsp-server/rtsp-client.c:
11400 * gst/rtsp-server/rtsp-session-media.c:
11401 * gst/rtsp-server/rtsp-session-media.h:
11402 * gst/rtsp-server/rtsp-stream-transport.c:
11403 * gst/rtsp-server/rtsp-stream-transport.h:
11404 rtsp: only create transport when needed
11405 Only create the StreamTransport when configured.
11407 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11409 * gst/rtsp-server/rtsp-client.c:
11410 client: small cleanup
11412 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11414 * gst/rtsp-server/rtsp-client.c:
11415 * gst/rtsp-server/rtsp-client.h:
11416 * gst/rtsp-server/rtsp-stream-transport.c:
11417 * gst/rtsp-server/rtsp-stream-transport.h:
11418 rtsp: refactor configuration of transport
11419 Move the configuration of the transport to a place where it makes
11422 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11424 * gst/rtsp-server/rtsp-client.c:
11425 client: refactor transport parsing
11427 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11429 * gst/rtsp-server/rtsp-client.c:
11430 client: refuse to change the MTU on shared media
11431 If we change the MTU of chared media, it changes for all clients.
11432 We don't want to set the MTU to something large for clients that
11435 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11437 * examples/test-mp4.c:
11438 * gst/rtsp-server/rtsp-media.c:
11439 small fixes to docs and debug
11441 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11443 * gst/rtsp-server/rtsp-stream.c:
11444 stream: transports must already have been removed
11446 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11448 * gst/rtsp-server/rtsp-media.c:
11449 * gst/rtsp-server/rtsp-stream.c:
11450 * gst/rtsp-server/rtsp-stream.h:
11451 stream: improve join and leave of the pipeline
11453 Do the cleanup properly
11456 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11458 * gst/rtsp-server/rtsp-media.c:
11459 media: move unprepare below default implementation
11460 Makes it easier to find the default implementation
11462 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11464 * gst/rtsp-server/rtsp-media.c:
11465 media: signal unprepared when we actually finish
11467 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11469 * gst/rtsp-server/rtsp-media.c:
11470 media: no need to unlock, unprepare does that when needed
11472 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11474 * docs/libs/gst-rtsp-server-sections.txt:
11475 * gst/rtsp-server/rtsp-media-factory.h:
11476 * gst/rtsp-server/rtsp-media-mapping.c:
11477 * gst/rtsp-server/rtsp-media.h:
11478 * gst/rtsp-server/rtsp-params.c:
11479 * gst/rtsp-server/rtsp-server.c:
11480 * gst/rtsp-server/rtsp-session-pool.h:
11481 * gst/rtsp-server/rtsp-session.c:
11482 * gst/rtsp-server/rtsp-session.h:
11483 * gst/rtsp-server/rtsp-stream-transport.h:
11484 * gst/rtsp-server/rtsp-stream.h:
11487 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11489 * gst/rtsp-server/rtsp-client.c:
11490 * gst/rtsp-server/rtsp-media-mapping.h:
11491 * gst/rtsp-server/rtsp-media.c:
11492 * gst/rtsp-server/rtsp-media.h:
11493 * gst/rtsp-server/rtsp-server.h:
11494 * gst/rtsp-server/rtsp-stream.c:
11495 * gst/rtsp-server/rtsp-stream.h:
11496 rtsp: fix MTU setting
11497 Fix setting of the MTU. There is no need for a vmethod.
11499 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11504 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11507 configure: bump version number after refactoring
11509 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11511 * gst/rtsp-server/Makefile.am:
11512 * gst/rtsp-server/rtsp-client.c:
11513 * gst/rtsp-server/rtsp-client.h:
11514 * gst/rtsp-server/rtsp-media-factory-uri.c:
11515 * gst/rtsp-server/rtsp-media-factory.c:
11516 * gst/rtsp-server/rtsp-media-factory.h:
11517 * gst/rtsp-server/rtsp-media.c:
11518 * gst/rtsp-server/rtsp-media.h:
11519 * gst/rtsp-server/rtsp-sdp.c:
11520 * gst/rtsp-server/rtsp-session-media.c:
11521 * gst/rtsp-server/rtsp-session-media.h:
11522 * gst/rtsp-server/rtsp-session.c:
11523 * gst/rtsp-server/rtsp-session.h:
11524 * gst/rtsp-server/rtsp-stream-transport.c:
11525 * gst/rtsp-server/rtsp-stream-transport.h:
11526 * gst/rtsp-server/rtsp-stream.c:
11527 * gst/rtsp-server/rtsp-stream.h:
11528 rtsp: massive refactoring
11529 Make GObjects from the remaining simple structures.
11530 Remove GstRTSPSessionStream, it's not needed.
11531 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
11532 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
11533 a GstRTSPStream should be transported to a client.
11534 Rename GstRTSPMediaFactory::get_element -> create_element because that
11535 more accurately describes what it does.
11536 Make nice methods instead of poking in the structures.
11537 Move some methods inside the relevant object source code.
11538 Use GPtrArray to store objects instead of plain arrays, it is more
11539 natural and allows us to more easily clean up.
11540 Move the allocation of udp ports to the Stream object. The Stream object
11541 contains the elements needed to stream the media to a client.
11542 Improve the prepare and unprepare methods. Unprepare should now undo
11543 everything prepare did. Improve also async unprepare when doing EOS on
11544 shutdown. Make sure we always unprepare correctly.
11546 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
11548 * gst/rtsp-server/rtsp-client.c:
11549 rtsp-client: Unref server address clients connected to
11550 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
11552 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
11554 * gst/rtsp-server/rtsp-server.c:
11555 rtsp-server: don't ref server socket if it is NULL
11556 Fixes test_bind_already_in_use unit test again after commit 6a497440.
11557 https://bugzilla.gnome.org/show_bug.cgi?id=686644
11559 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
11561 * tests/check/Makefile.am:
11562 tests: Add libgio link dependency
11563 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
11565 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11567 * gst/rtsp-server/rtsp-media-mapping.c:
11568 * gst/rtsp-server/rtsp-media-mapping.h:
11569 rtsp-media-mapping: rename find_media vfunc to find_factory
11570 The virtual method and class method should have the same name
11571 so it is correctly represented in GIR file
11572 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11574 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11576 * gst/rtsp-server/rtsp-auth.c:
11577 * gst/rtsp-server/rtsp-client.c:
11578 * gst/rtsp-server/rtsp-media-factory-uri.c:
11579 * gst/rtsp-server/rtsp-media-factory.c:
11580 * gst/rtsp-server/rtsp-media-mapping.c:
11581 * gst/rtsp-server/rtsp-media.c:
11582 * gst/rtsp-server/rtsp-server.c:
11583 * gst/rtsp-server/rtsp-session-pool.c:
11584 * gst/rtsp-server/rtsp-session.c:
11585 rtsp-server: fixed comments and GIR annotations
11586 https://bugzilla.gnome.org/show_bug.cgi?id=680777
11588 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11590 * gst/rtsp-server/rtsp-media-mapping.c:
11591 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
11593 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
11595 * gst/rtsp-server/rtsp-server.c:
11596 rtsp-server: allow binding on port 0 (binds on a random port)
11598 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11600 * gst/rtsp-server/rtsp-server.c:
11601 * gst/rtsp-server/rtsp-server.h:
11602 rtsp-server: add bound-port property
11603 bound-port can be used to retrieve the port number when the server is bound on
11604 port 0, which binds on a random port.
11606 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11608 * gst/rtsp-server/rtsp-media-factory.c:
11609 * gst/rtsp-server/rtsp-media-factory.h:
11610 rtsp-media-factory: make ::get_element overridable by GI bindings
11611 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11612 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11613 as the invoker for ::get_element(), making it overridable by GI generated
11616 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11618 * gst/rtsp-server/rtsp-media-factory-uri.c:
11619 rtsp-media-factory-uri: don't autoplug parsers in a loop
11620 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11623 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11625 * gst/rtsp-server/Makefile.am:
11626 Explicitly link against gio. Fix link error on mac.
11628 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11630 * gst/rtsp-server/rtsp-session.c:
11631 session: add ttl to the transport header in SETUP
11632 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11634 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11636 * gst/rtsp-server/rtsp-client.c:
11637 * gst/rtsp-server/rtsp-client.h:
11638 * gst/rtsp-server/rtsp-media.c:
11639 client: Use client transport settings for multicast if allowed.
11640 This patch makes it possible for the client to send transport settings for
11641 multicast (destination && ttl). Client settings must be explicitly allowed or
11642 the server will use its own settings.
11643 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11645 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11648 Automatic update of common submodule
11649 From 6c0b52c to 6bb6951
11651 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11653 * gst/rtsp-server/rtsp-client.c:
11654 rtsp-client: do not destroy the rtsp watch
11655 Don't destroy the client watch while dispatching. The rtsp watch is
11656 automatically destroyed after the rtsp watch function closed() has
11658 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11660 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11663 Automatic update of common submodule
11664 From 4f962f7 to 6c0b52c
11666 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11668 * gst/rtsp-server/rtsp-media.c:
11669 media: fix check for seekability
11671 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11673 * gst/rtsp-server/rtsp-client.c:
11674 client: use more GIO
11675 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11677 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11679 * gst/rtsp-server/rtsp-server.c:
11680 server: remove obsolete includes
11682 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11684 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11685 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11686 be available in "on_new_ssrc". The transports are added in
11687 gst_rtsp_media_set_state when going to PLAYING state. However,
11688 "on_new_ssrc" might be called before this happens.
11689 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11691 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11693 * gst/rtsp-server/rtsp-client.c:
11694 * gst/rtsp-server/rtsp-client.h:
11695 rtsp-client: add signals for rtsp requests (fixes #683287)
11697 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11699 * gst/rtsp-server/rtsp-client.c:
11700 * gst/rtsp-server/rtsp-client.h:
11701 add new-session signal to rtsp-client (fixes #683058)
11703 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11706 Automatic update of common submodule
11707 From 668acee to 4f962f7
11709 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11711 * gst/rtsp-server/rtsp-server.c:
11712 * tests/check/gst/rtspserver.c:
11713 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11714 Do not assume that *error is set in g_socket_address_enumerator_next.
11715 Added test_bind_already_in_use unit-test.
11716 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11718 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11721 Automatic update of common submodule
11722 From 94ccf4c to 668acee
11724 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11726 * gst/rtsp-server/rtsp-client.c:
11727 * gst/rtsp-server/rtsp-client.h:
11728 rtsp-client: make create_sdp virtual method
11729 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11731 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11734 Automatic update of common submodule
11735 From 98e386f to 94ccf4c
11737 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11739 * gst/rtsp-server/rtsp-client.c:
11742 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11744 * gst/rtsp-server/rtsp-client.c:
11745 * gst/rtsp-server/rtsp-client.h:
11746 * gst/rtsp-server/rtsp-server.c:
11747 * gst/rtsp-server/rtsp-server.h:
11748 rtsp-server: use an existing socket to establish HTTP tunnel
11749 Make it possible to transfer a socket from an HTTP server to be used as
11750 an RTSP over HTTP tunnel.
11752 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11754 * gst/rtsp-server/rtsp-client.c:
11755 * gst/rtsp-server/rtsp-media.c:
11756 * gst/rtsp-server/rtsp-media.h:
11757 rtsp: Handle the blocksize parameter
11758 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11760 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11762 * tests/check/Makefile.am:
11763 * tests/check/gst/rtspserver.c:
11764 Have unit test get header from source dir, not installed dir
11765 This makes compilation of unit tests work in a build directory other
11766 than the source directory.
11767 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11769 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11771 * gst/rtsp-server/rtsp-media.c:
11772 rtsp-media: update for gst_element_make_from_uri() changes
11774 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11777 * tests/Makefile.am:
11778 * tests/check/Makefile.am:
11779 * tests/check/gst/rtspserver.c:
11780 rtsp: add unit test
11781 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11783 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11785 * gst/rtsp-server/rtsp-media.c:
11786 rtsp-media: don't collect media stats when going to NULL
11787 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11789 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11791 * gst/rtsp-server/rtsp-client.c:
11792 client: don't leak transports
11794 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11796 * gst/rtsp-server/rtsp-client.c:
11797 rtsp-client: free transport on no_stream in SETUP handler
11799 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11801 * gst/rtsp-server/rtsp-client.c:
11802 rtsp-client: changed session media iteration
11803 In client_unlink_session: now don't iterate in session->medias
11804 list where items are removed by gst_rtsp_session_release_media.
11805 Instead, repeatedly remove the first item.
11807 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11809 * gst/rtsp-server/rtsp-client.c:
11810 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11811 GstRTSPSessionMedia is not a GObject type. When the
11812 GstRTSPSession is freed, it will free the media.
11814 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11816 * gst/rtsp-server/rtsp-media-factory.c:
11817 factory: plug pad leak in collect_streams
11818 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11819 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11820 will take one reference, and the other reference will otherwise
11821 give a memory leak.
11823 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11826 configure: suppress some warnings when debug is disabled
11827 Warnings about unused variables should be suppressed if core has the
11828 debug system disabled.
11829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11831 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11833 * docs/libs/Makefile.am:
11834 docs: fix build in uninstalled setup
11835 Include gst-plugins-base libs properly.
11837 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11839 * docs/libs/gst-rtsp-server.types:
11840 docs: include headers defining rtsp-server object types
11841 Fixes compiler warnings during docs build.
11842 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11844 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11847 configure: Add warning flags for compiler when configuring
11848 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11850 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11853 Automatic update of common submodule
11854 From 03a0e57 to 98e386f
11856 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11859 Automatic update of common submodule
11860 From 1fab359 to 03a0e57
11862 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11864 * gst/rtsp-server/rtsp-client.c:
11865 client: fix GSocketAddress leak in gst_rtsp_client_accept
11866 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11868 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11871 Automatic update of common submodule
11872 From f1b5a96 to 1fab359
11874 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11877 Automatic update of common submodule
11878 From 92b7266 to f1b5a96
11880 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11883 Automatic update of common submodule
11884 From ec1c4a8 to 92b7266
11886 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11889 Automatic update of common submodule
11890 From 3429ba6 to ec1c4a8
11892 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11894 * gst/rtsp-server/rtsp-auth.c:
11895 * gst/rtsp-server/rtsp-client.c:
11896 * gst/rtsp-server/rtsp-media-factory-uri.c:
11897 * gst/rtsp-server/rtsp-server.c:
11898 rtsp: fix compiler warnings
11899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11901 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11904 Automatic update of common submodule
11905 From dc70203 to 3429ba6
11907 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11909 * gst/rtsp-server/rtsp-client.c:
11910 * gst/rtsp-server/rtsp-media-factory.c:
11911 * gst/rtsp-server/rtsp-media-factory.h:
11912 * gst/rtsp-server/rtsp-media.c:
11913 * gst/rtsp-server/rtsp-media.h:
11914 * gst/rtsp-server/rtsp-server.c:
11915 * gst/rtsp-server/rtsp-server.h:
11916 * gst/rtsp-server/rtsp-session-pool.c:
11917 * gst/rtsp-server/rtsp-session-pool.h:
11918 rtsp-server: port to new thread API
11920 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11923 Automatic update of common submodule
11924 From 6db25be to dc70203
11926 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11928 * gst/rtsp-server/rtsp-auth.c:
11929 * gst/rtsp-server/rtsp-auth.h:
11930 * gst/rtsp-server/rtsp-client.c:
11931 rtsp-server: Fix compilation and compiler warnings
11933 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11937 * gst/rtsp-server/Makefile.am:
11938 configure: Modernize autotools setup a bit
11939 Also we now only create tar.bz2 and tar.xz tarballs.
11941 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11944 Automatic update of common submodule
11945 From 464fe15 to 6db25be
11947 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11950 Automatic update of common submodule
11951 From 7fda524 to 464fe15
11953 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11956 * docs/libs/Makefile.am:
11957 * docs/version.entities.in:
11958 * gst-rtsp.spec.in:
11959 * gst/rtsp-server/Makefile.am:
11960 * pkgconfig/Makefile.am:
11961 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11962 * pkgconfig/gstreamer-rtsp-server.pc.in:
11963 * tests/Makefile.am:
11964 rtsp-server: Update versioning
11966 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11968 Merge remote-tracking branch 'origin/0.10'
11970 gst/rtsp-server/rtsp-session-pool.c
11972 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11974 * gst/rtsp-server/rtsp-session-pool.c:
11975 rtsp-server: Don't use deprecated GLib API
11977 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11979 Replace master with 0.11
11981 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11983 Merge branch 'master' into 0.11
11985 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11987 Merge branch 'master' into 0.11
11989 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11992 A couple minor typo fixes
11994 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11996 * gst/rtsp-server/rtsp-media.c:
11997 media: fix state of the appqueue
11999 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12001 * gst/rtsp-server/rtsp-media-factory-uri.c:
12002 factory: use videoconvert
12004 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12006 * gst/rtsp-server/rtsp-media-factory-uri.c:
12007 factory: change to new style caps
12009 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12011 * gst/rtsp-server/rtsp-client.c:
12012 * gst/rtsp-server/rtsp-client.h:
12013 * gst/rtsp-server/rtsp-media-factory-uri.c:
12014 * gst/rtsp-server/rtsp-media.c:
12015 * gst/rtsp-server/rtsp-server.c:
12016 * gst/rtsp-server/rtsp-server.h:
12017 * gst/rtsp-server/rtsp-session-pool.c:
12018 rtsp-server: port to GIO
12021 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12024 configure: fix build
12026 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12029 docs: fix for gst_rtsp_server_set_port() -> _set_service()
12030 https://bugzilla.gnome.org/show_bug.cgi?id=666548
12032 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12035 * examples/Makefile.am:
12036 First rule of gst-rtsp-server club: don't talk about gst-phonon
12038 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12041 * pkgconfig/Makefile.am:
12042 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
12043 * pkgconfig/gstreamer-rtsp-server.pc.in:
12044 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
12045 For consistency with all other modules.
12047 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12049 * gst/rtsp-server/rtsp-client.c:
12050 rtsp-client: update for new map API
12052 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12055 * bindings/Makefile.am:
12056 * bindings/python/Makefile.am:
12057 * bindings/python/arg-types.py:
12058 * bindings/python/codegen/Makefile.am:
12059 * bindings/python/codegen/__init__.py:
12060 * bindings/python/codegen/argtypes.py:
12061 * bindings/python/codegen/code-coverage.py:
12062 * bindings/python/codegen/codegen.py:
12063 * bindings/python/codegen/definitions.py:
12064 * bindings/python/codegen/defsparser.py:
12065 * bindings/python/codegen/docextract.py:
12066 * bindings/python/codegen/docgen.py:
12067 * bindings/python/codegen/fileprefix.override:
12068 * bindings/python/codegen/fileprefixmodule.c:
12069 * bindings/python/codegen/h2def.py:
12070 * bindings/python/codegen/mergedefs.py:
12071 * bindings/python/codegen/mkskel.py:
12072 * bindings/python/codegen/override.py:
12073 * bindings/python/codegen/reversewrapper.py:
12074 * bindings/python/codegen/scmexpr.py:
12075 * bindings/python/rtspserver-types.defs:
12076 * bindings/python/rtspserver.defs:
12077 * bindings/python/rtspserver.override:
12078 * bindings/python/rtspservermodule.c:
12079 * bindings/python/test.py:
12081 python: remove pygst-based python bindings
12082 pygi is the future, apparently.
12084 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
12087 Automatic update of common submodule
12088 From c463bc0 to 7fda524
12090 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12093 Automatic update of common submodule
12094 From 2a59016 to c463bc0
12096 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12099 Automatic update of common submodule
12100 From 0807187 to 2a59016
12102 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12105 Automatic update of common submodule
12106 From 11f0cd5 to 0807187
12108 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12110 * examples/test-auth.c:
12111 example: update for new caps
12113 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12115 * examples/test-video.c:
12116 * gst/rtsp-server/rtsp-client.c:
12117 * gst/rtsp-server/rtsp-media-factory-uri.c:
12118 * gst/rtsp-server/rtsp-media.c:
12119 * gst/rtsp-server/rtsp-media.h:
12120 * gst/rtsp-server/rtsp-session.c:
12121 * gst/rtsp-server/rtsp-session.h:
12122 rtsp-server: port some more to 0.11
12124 Remove bufferlist stuff
12125 Update for new API.
12126 Add queue before appsink now that preroll-queue-len is gone.
12127 Update for request pad changes.
12129 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12131 Merge branch 'master' into 0.11
12133 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12135 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12136 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12137 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12139 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
12141 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12142 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
12143 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12145 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12147 Merge branch 'master' into 0.11
12149 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12151 * gst/rtsp-server/rtsp-media.c:
12152 * gst/rtsp-server/rtsp-media.h:
12153 media: add a seekable boolean
12154 Maintain the seekable state with a new variable instead of reusing the
12157 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
12159 * gst/rtsp-server/rtsp-media.c:
12160 Disallow seek in live media
12162 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12164 Merge branch 'master' into 0.11
12166 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
12168 * gst/rtsp-server/rtsp-server.c:
12169 #ifdef statements for windows socket creation were missing
12171 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
12174 Automatic update of common submodule
12175 From a39eb83 to 11f0cd5
12177 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
12180 Automatic update of common submodule
12181 From 605cd9a to a39eb83
12183 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12185 Merge branch 'master' into 0.11
12187 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12189 * gst/rtsp-server/rtsp-client.c:
12190 client: use method to access property
12192 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12194 * gst/rtsp-server/rtsp-media-factory.c:
12195 * gst/rtsp-server/rtsp-media-factory.h:
12196 media-factory: add protocols property
12197 Add a property to configure the allowed protocols in the media created from the
12200 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12202 * gst/rtsp-server/rtsp-media-factory.c:
12203 * gst/rtsp-server/rtsp-media-factory.h:
12204 media-factory: add media-configure signal
12205 Add signal to allow the application to configure the media after it was created
12208 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12210 * gst/rtsp-server/rtsp-client.c:
12211 client: use method to access property
12213 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12215 * gst/rtsp-server/rtsp-media-factory.c:
12216 * gst/rtsp-server/rtsp-media-factory.h:
12217 media-factory: add protocols property
12218 Add a property to configure the allowed protocols in the media created from the
12221 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12223 * gst/rtsp-server/rtsp-media-factory.c:
12224 * gst/rtsp-server/rtsp-media-factory.h:
12225 media-factory: add media-configure signal
12226 Add signal to allow the application to configure the media after it was created
12229 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12231 Merge branch 'master' into 0.11
12233 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12235 * gst/rtsp-server/rtsp-client.c:
12236 client: use media multicast group
12238 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12240 * gst/rtsp-server/rtsp-media-factory.h:
12241 * gst/rtsp-server/rtsp-server.h:
12242 * gst/rtsp-server/rtsp-session-pool.h:
12243 * gst/rtsp-server/rtsp-session.h:
12246 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12248 * gst/rtsp-server/rtsp-client.c:
12249 * gst/rtsp-server/rtsp-sdp.h:
12250 sdp: copy and free the server ip address
12251 Copy and free the server ip address to make memory management easier later.
12253 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12255 * gst/rtsp-server/rtsp-media-factory.c:
12256 media-factory: configure multicast in media
12258 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12260 * gst/rtsp-server/rtsp-media.c:
12261 * gst/rtsp-server/rtsp-media.h:
12262 media: add property for multicast group
12263 Add a property to configure the multicast group in the media.
12264 Based on patches from Marc Leeman and Robert Krakora.
12266 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12268 * gst/rtsp-server/rtsp-media-factory.c:
12269 * gst/rtsp-server/rtsp-media-factory.h:
12270 media-factory: add property for multicast group
12271 Add a property to configure the multicast group in the media factory.
12272 Based on patches from Marc Leeman and Robert Krakora.
12274 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12276 * gst/rtsp-server/rtsp-client.c:
12277 client: do configuration of transport in one place
12278 Move the configuration of the transport destination address to where we also
12279 configure the other bits.
12281 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12283 * gst/rtsp-server/rtsp-client.c:
12284 client: use media multicast group
12286 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12288 * gst/rtsp-server/rtsp-media-factory.h:
12289 * gst/rtsp-server/rtsp-server.h:
12290 * gst/rtsp-server/rtsp-session-pool.h:
12291 * gst/rtsp-server/rtsp-session.h:
12294 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12296 * gst/rtsp-server/rtsp-client.c:
12297 * gst/rtsp-server/rtsp-sdp.h:
12298 sdp: copy and free the server ip address
12299 Copy and free the server ip address to make memory management easier later.
12301 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12303 * gst/rtsp-server/rtsp-media-factory.c:
12304 media-factory: configure multicast in media
12306 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12308 * gst/rtsp-server/rtsp-media.c:
12309 * gst/rtsp-server/rtsp-media.h:
12310 media: add property for multicast group
12311 Add a property to configure the multicast group in the media.
12312 Based on patches from Marc Leeman and Robert Krakora.
12314 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12316 * gst/rtsp-server/rtsp-media-factory.c:
12317 * gst/rtsp-server/rtsp-media-factory.h:
12318 media-factory: add property for multicast group
12319 Add a property to configure the multicast group in the media factory.
12320 Based on patches from Marc Leeman and Robert Krakora.
12322 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12324 * gst/rtsp-server/rtsp-client.c:
12325 client: do configuration of transport in one place
12326 Move the configuration of the transport destination address to where we also
12327 configure the other bits.
12329 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12331 Merge branch 'master' into 0.11
12333 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12335 * gst/rtsp-server/rtsp-client.c:
12336 client: destroy pipeline on client disconnect with no prior TEARDOWN.
12337 The problem occurs when the client abruptly closes the connection without
12338 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
12339 server is where the pipeline gets torn down. Since this handler is not called,
12340 the pipeline remains and is up and running. Subsequent clients get their own
12341 pipelines and if the do not issue TEARDOWNs then those pipelines will also
12342 remain up and running. This is a resource leak.
12344 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12346 Merge branch 'master' into 0.11
12348 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
12350 * gst/rtsp-server/rtsp-media-factory.c:
12351 * gst/rtsp-server/rtsp-media-factory.h:
12352 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
12353 For example, it can be used to retrieve source elements like appsrc, in a more
12354 convenient way than subclassing get_element.
12356 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12358 Merge branch 'master' into 0.11
12360 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
12362 * gst/rtsp-server/rtsp-server.c:
12363 rtsp-server: hold on to reference while using object
12365 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12367 * gst/rtsp-server/rtsp-media.c:
12370 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12373 configure: use unstable api
12375 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
12377 * gst/rtsp-server/rtsp-client.c:
12378 client: fix reference counting
12380 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
12382 * gst/rtsp-server/rtsp-client.c:
12383 * gst/rtsp-server/rtsp-media.c:
12384 fix compiler warnings about unused variables
12386 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
12388 * examples/test-launch.c:
12389 * examples/test-readme.c:
12390 * examples/test-uri.c:
12391 * examples/test-video.c:
12392 examples: tell rtsp uri when ready
12394 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
12397 Automatic update of common submodule
12398 From 69b981f to 605cd9a
12400 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12402 * gst/rtsp-server/rtsp-client.c:
12403 client: update for buffer API change
12405 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12407 * gst/rtsp-server/Makefile.am:
12408 Makefile.am: 0.10 => @GST_MAJORMINOR@
12410 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12412 * gst/rtsp-server/rtsp-media-factory-uri.c:
12413 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
12415 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12417 * gst/rtsp-server/.gitignore:
12418 .gitignore: 0.10 => 0.11
12420 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
12422 * gst/rtsp-server/Makefile.am:
12423 Makefile.am: 0.10 => @GST_MAJORMINOR@
12425 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12427 Merge branch 'master' into 0.11
12429 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
12432 Automatic update of common submodule
12433 From 9e5bbd5 to 69b981f
12435 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
12438 Automatic update of common submodule
12439 From fd35073 to 9e5bbd5
12441 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
12444 Automatic update of common submodule
12445 From 46dfcea to fd35073
12447 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12449 * gst/rtsp-server/rtsp-media-factory-uri.c:
12450 * gst/rtsp-server/rtsp-media.c:
12451 media: port to new caps API
12453 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12455 Merge branch 'master' into 0.11
12457 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12459 * bindings/vala/gst-rtsp-server-0.10.vapi:
12460 Updated Vala bindings.
12461 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12463 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
12465 * gst/rtsp-server/rtsp-server.c:
12466 * gst/rtsp-server/rtsp-server.h:
12467 Add a signal for newly connected clients.
12468 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
12470 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
12472 * bindings/python/rtspserver.override:
12473 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
12475 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12477 * gst/rtsp-server/Makefile.am:
12478 * gst/rtsp-server/rtsp-client.c:
12479 * gst/rtsp-server/rtsp-funnel.c:
12480 * gst/rtsp-server/rtsp-funnel.h:
12481 * gst/rtsp-server/rtsp-media.c:
12482 rtsp-server: port to 0.11
12484 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12489 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12491 Merge branch 'master' into 0.11
12496 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12499 Automatic update of common submodule
12500 From c3cafe1 to 46dfcea
12502 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
12504 * bindings/python/Makefile.am:
12505 * bindings/python/rtspserver.defs:
12506 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
12508 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
12510 * bindings/python/arg-types.py:
12511 python bindings: add GstRTSPUrlParam
12512 Needed to implement MediaFactory virtual proxies
12514 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
12516 * bindings/python/arg-types.py:
12517 python bindings: fix returning GstRTSPUrl types
12519 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
12521 * bindings/python/arg-types.py:
12522 python bindings: add arg type for GstRTSPUrl
12524 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
12526 * bindings/python/rtspserver.defs:
12527 python bindings: fix the definition of MediaFactory.collect_stream
12529 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
12532 Automatic update of common submodule
12533 From 1ccbe09 to c3cafe1
12535 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12538 Automatic update of common submodule
12539 From 193b717 to 1ccbe09
12541 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
12544 Automatic update of common submodule
12545 From b77e2bf to 193b717
12547 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12550 build: Include lcov.mak to allow test coverage report generation
12552 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12555 Automatic update of common submodule
12556 From d8814b6 to b77e2bf
12558 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12561 Automatic update of common submodule
12562 From 6aaa286 to d8814b6
12564 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
12567 Automatic update of common submodule
12568 From 6aec6b9 to 6aaa286
12570 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
12573 autogen: wingo signed comment
12575 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
12577 * gst/rtsp-server/rtsp-session-pool.c:
12578 session: use full charset for RTSP session ID
12579 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
12580 session ID more difficult.
12581 https://bugzilla.gnome.org/show_bug.cgi?id=643812
12583 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12585 * gst/rtsp-server/Makefile.am:
12586 rtsp-server: Don't install the funnel header
12588 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12591 Automatic update of common submodule
12592 From 1de7f6a to 6aec6b9
12594 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12597 configure: require core/base 0.10.31
12598 Needed at least for gst_plugin_feature_rank_compare_func().
12600 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12603 Automatic update of common submodule
12604 From f94d739 to 1de7f6a
12606 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12608 * gst/rtsp-server/rtsp-media.c:
12609 media: remove more unused code
12611 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12613 * gst/rtsp-server/rtsp-media.c:
12614 * gst/rtsp-server/rtsp-media.h:
12615 media: remove duplicate filtering
12616 Remove the duplicate filtering code now that we have a released -good version.
12617 Give a warning instead.
12619 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12621 * gst/rtsp-server/rtsp-media-factory.c:
12622 * gst/rtsp-server/rtsp-media.c:
12623 media: fix default buffer size
12625 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12627 * gst/rtsp-server/rtsp-media-factory.c:
12628 * gst/rtsp-server/rtsp-media-factory.h:
12629 media-factory: add property to configure the buffer-size
12630 Add a property to configure the kernel UDP buffer size.
12632 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12634 * gst/rtsp-server/rtsp-media.c:
12635 * gst/rtsp-server/rtsp-media.h:
12636 media: add property to configure kernel buffer sizes
12637 Add a property to configure the kernel UDP buffer size.
12639 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12642 configure: set PYGOBJECT_REQ before using it
12643 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12645 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12647 * docs/Makefile.am:
12648 docs: recursive into sub-directories on 'make upload'
12650 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12652 * docs/libs/gst-rtsp-server-docs.sgml:
12653 * docs/version.entities.in:
12654 docs: mention full version these docs are for, not just major-minor
12656 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12659 back to development
12661 === release 0.10.8 ===
12663 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12668 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12670 * gst/rtsp-server/rtsp-server.c:
12671 rtsp-server: clarify docs a little
12673 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12675 * gst/rtsp-server/rtsp-media.c:
12676 media: init debug category before starting thread
12678 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12680 * gst/rtsp-server/rtsp-auth.c:
12681 auth: add realm to make it more spec compliant
12683 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12685 * gst/rtsp-server/rtsp-server.c:
12686 * gst/rtsp-server/rtsp-server.h:
12687 server: add locking
12689 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12691 * examples/test-video.c:
12692 example: improve example docs a little
12694 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12696 * gst/rtsp-server/rtsp-server.c:
12697 server: ensure the watch has a ref to the server
12699 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12701 * gst/rtsp-server/rtsp-server.c:
12702 server: simpify channel function
12704 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12706 * gst/rtsp-server/rtsp-server.c:
12707 * gst/rtsp-server/rtsp-server.h:
12708 server: simplify management of channel and source
12709 We don't need to keep around the channel and source objects. Let the mainloop
12710 and the source manage the source and channel respectively.
12712 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12718 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12720 * tests/.gitignore:
12721 * tests/Makefile.am:
12722 * tests/test-cleanup.c:
12723 tests: add tests directory and cleanup test
12725 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12727 * gst/rtsp-server/rtsp-media-factory-uri.c:
12728 * gst/rtsp-server/rtsp-media-factory.c:
12729 * gst/rtsp-server/rtsp-media-mapping.c:
12730 * gst/rtsp-server/rtsp-media.c:
12731 * gst/rtsp-server/rtsp-session-pool.c:
12732 * gst/rtsp-server/rtsp-session.c:
12733 server: improve debugging in various objects
12735 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12737 * gst/rtsp-server/rtsp-server.c:
12738 server: chain up to the parent finalize
12740 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12742 * bindings/python/rtspserver-types.defs:
12743 * bindings/python/rtspserver.defs:
12744 * bindings/python/rtspserver.override:
12745 * bindings/python/test.py:
12746 gst-rtsp-server: update python bindings
12748 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12750 * gst/rtsp-server/rtsp-client.c:
12751 client: use the response from the clientstate
12752 Create the response object only once and store in the client state.
12753 Make all methods use the state response,
12755 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12757 * gst/rtsp-server/rtsp-server.c:
12758 server: use signal to keep track of clients
12759 Keep track of all the clients that the server creates and remove them when they
12760 fire the 'closed' signal.
12762 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12764 * gst/rtsp-server/rtsp-client.c:
12765 * gst/rtsp-server/rtsp-client.h:
12766 client: emit signal when closing
12768 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12770 * examples/.gitignore:
12771 * examples/Makefile.am:
12772 * examples/test-auth.c:
12773 * examples/test-video.c:
12774 * gst/rtsp-server/rtsp-auth.c:
12775 * gst/rtsp-server/rtsp-auth.h:
12776 * gst/rtsp-server/rtsp-client.c:
12777 * gst/rtsp-server/rtsp-media-factory.c:
12778 * gst/rtsp-server/rtsp-media.c:
12779 * gst/rtsp-server/rtsp-media.h:
12780 * gst/rtsp-server/rtsp-session-pool.h:
12781 * gst/rtsp-server/rtsp-session.h:
12782 media: enable per factory authorisations
12783 Allow for adding a GstRTSPAuth on the factory and media level and check
12784 permissions when accessing the factory.
12785 Add hints to the auth methods for future more fine grained authorisation.
12786 Add example application for per factory authentication.
12788 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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-client.h:
12794 * gst/rtsp-server/rtsp-params.c:
12795 * gst/rtsp-server/rtsp-params.h:
12796 rtsp-server: Pass ClientState structure arround
12797 Pass the collected information for the ongoing request in a GstRTSPClientState
12798 structure that we can then pass around to simplify the method arguments. This
12799 will also be handy when we implement logging functionality.
12801 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12803 * gst/rtsp-server/rtsp-media-factory.c:
12804 * gst/rtsp-server/rtsp-media-factory.h:
12805 media-factory: add methods to configure authorisation
12807 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12809 * gst/rtsp-server/rtsp-client.c:
12810 client: unref auth in finalize
12812 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12814 * gst/rtsp-server/rtsp-server.c:
12815 server: unref auth in finalize
12817 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12819 * docs/libs/gst-rtsp-server-docs.sgml:
12820 * docs/libs/gst-rtsp-server-sections.txt:
12821 * docs/libs/gst-rtsp-server.types:
12822 docs: add more docs
12824 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12826 * gst/rtsp-server/rtsp-server.c:
12827 * gst/rtsp-server/rtsp-server.h:
12828 server: separate create and accept
12829 Create separate create and accept methods so that subclasses can create custom
12831 Configure the server in the client object and prepare for keeping track of
12834 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12836 * gst/rtsp-server/rtsp-client.c:
12837 * gst/rtsp-server/rtsp-client.h:
12838 client: add support for setting the server.
12839 Add support for keeping a ref to the server that started this client
12842 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12844 * gst/rtsp-server/rtsp-auth.c:
12845 auth: fix memleak and add some docs
12846 Fix a memleak of the basic auth token.
12847 Add docs for the helper function
12849 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12851 * gst/rtsp-server/rtsp-auth.c:
12852 * gst/rtsp-server/rtsp-auth.h:
12853 * gst/rtsp-server/rtsp-client.c:
12854 client: delegate setup of auth to the manager
12855 Delegate the configuration of the authentication tokens to the manager object
12858 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12860 * examples/test-video.c:
12861 * gst/rtsp-server/Makefile.am:
12862 * gst/rtsp-server/rtsp-auth.c:
12863 * gst/rtsp-server/rtsp-auth.h:
12864 * gst/rtsp-server/rtsp-client.c:
12865 * gst/rtsp-server/rtsp-client.h:
12866 * gst/rtsp-server/rtsp-server.c:
12867 * gst/rtsp-server/rtsp-server.h:
12868 auth: add authentication object
12869 Add an object that can check the authorization of requests.
12870 Implement basic authentication.
12871 Add example authentication to test-video
12873 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12875 * gst/rtsp-server/rtsp-server.c:
12876 * gst/rtsp-server/rtsp-server.h:
12877 server: move includes back
12878 the includes are needed for sockaddr_in.
12880 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12882 * gst/rtsp-server/rtsp-client.c:
12883 * gst/rtsp-server/rtsp-client.h:
12884 * gst/rtsp-server/rtsp-server.c:
12885 * gst/rtsp-server/rtsp-server.h:
12886 rtsp: move network includes where they are needed
12888 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12890 * gst/rtsp-server/rtsp-media.h:
12891 rtsp-media.h: Minor corrections in comments.
12894 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12897 Automatic update of common submodule
12898 From e572c87 to f94d739
12900 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12904 * docs/libs/.gitignore:
12905 * examples/.gitignore:
12906 * gst/rtsp-server/.gitignore:
12909 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12911 * docs/libs/Makefile.am:
12912 docs: We don't build ps/pdf for API reference docs
12914 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12917 Automatic update of common submodule
12918 From ccbaa85 to e572c87
12920 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12923 Automatic update of common submodule
12924 From 46445ad to ccbaa85
12926 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12928 * gst/rtsp-server/Makefile.am:
12929 * gst/rtsp-server/rtsp-funnel.c:
12930 * gst/rtsp-server/rtsp-funnel.h:
12931 * gst/rtsp-server/rtsp-media.c:
12932 funnel: rename fsfunnel to rtspfunnel
12933 Rename the funnel to avoid conflicts with the farsight one.
12935 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12937 * gst/rtsp-server/Makefile.am:
12938 * gst/rtsp-server/fs-funnel.c:
12939 * gst/rtsp-server/fs-funnel.h:
12940 * gst/rtsp-server/rtsp-media.c:
12941 rtsp-media: add and use fsfunnel
12942 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12943 select-all property that we need.
12945 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12947 * gst/rtsp-server/Makefile.am:
12948 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12949 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12950 for the g-ir-compiler, rather than just assuming the env var has
12953 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12960 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12962 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12965 * gst/rtsp-server/Makefile.am:
12966 gobject-introspection: fix g-i build for uninstalled setup
12967 Requires gst-plugins-base git (> 0.10.31.2).
12969 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12971 * examples/test-uri.c:
12972 examples: add some more options and comments
12974 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12976 * gst/rtsp-server/rtsp-media-factory-uri.c:
12977 factory-uri: use right property type
12979 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12981 * gst/rtsp-server/rtsp-media-factory-uri.c:
12982 factory-uri: attempt to configure buffer-lists
12983 Attempt to configure buffer lists in the payloader for improved performance.
12985 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12987 * gst/rtsp-server/rtsp-media.c:
12988 media: attempt to configure bigger UDP buffers
12989 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12990 send buffers with high bitrate streams.
12992 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12994 * gst/rtsp-server/rtsp-client.c:
12995 client: use the socket length from getsockname
12996 Use the length returned by getsockname to perform the getnameinfo call because
12997 the size can depend on the socket type and platform.
13000 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13002 * docs/libs/gst-rtsp-server-docs.sgml:
13003 * docs/libs/gst-rtsp-server-sections.txt:
13004 docs: add uri factory to the docs
13006 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13008 * gst/rtsp-server/rtsp-client.c:
13009 * gst/rtsp-server/rtsp-media.h:
13012 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13014 * gst/rtsp-server/rtsp-client.c:
13015 * gst/rtsp-server/rtsp-media.c:
13016 * gst/rtsp-server/rtsp-media.h:
13017 * gst/rtsp-server/rtsp-session.c:
13018 * gst/rtsp-server/rtsp-session.h:
13019 rtsp-server: add support for buffer lists
13020 Add support for sending bufferlists received from appsink.
13023 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13025 * gst/rtsp-server/rtsp-client.c:
13026 * gst/rtsp-server/rtsp-media.c:
13027 * gst/rtsp-server/rtsp-media.h:
13028 * gst/rtsp-server/rtsp-sdp.c:
13029 media: make method to retrieve the play range
13030 Make a method to retrieve the playback range so that we can conditionally create
13031 a different range for the SDP and the PLAY requests.
13033 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13035 * gst/rtsp-server/rtsp-media.c:
13036 * gst/rtsp-server/rtsp-media.h:
13037 media: add signal to notify of state changes
13039 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13041 * gst/rtsp-server/rtsp-client.h:
13042 client: cleanup headers
13044 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13046 * gst/rtsp-server/rtsp-client.c:
13049 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13051 * gst/rtsp-server/rtsp-media-factory-uri.c:
13052 * gst/rtsp-server/rtsp-media-factory-uri.h:
13053 factory-uri: add support for gstpay
13054 Add an option to prefer gstpay over decoder + raw payloader.
13056 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13058 * gst/rtsp-server/rtsp-media-factory-uri.c:
13059 * gst/rtsp-server/rtsp-media-factory-uri.h:
13060 factory-uri: rework the autoplugger.
13061 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
13064 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13066 * gst/rtsp-server/rtsp-media-factory-uri.c:
13067 factory-uri: use better factory filter
13068 Make better payloader filter based on autoplug rank and RTP use case.
13070 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13073 Automatic update of common submodule
13074 From 169462a to 46445ad
13076 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13078 * gst/rtsp-server/rtsp-server.c:
13079 server: set SO_REUSEADDR before bind
13080 Set the SO_REUSEADDR _before_ bind() to make it actually work.
13082 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13084 * gst/rtsp-server/rtsp-media.c:
13085 * gst/rtsp-server/rtsp-media.h:
13086 media: emit prepared signal when prepared
13087 Make a 'prepared' signal and emit it when we successfully prepared the element.
13088 This signal can be used to configure the media object after it has been prepared
13091 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
13094 Automatic update of common submodule
13095 From 011bcc8 to 169462a
13097 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
13099 python an optional dependency
13100 * configure.ac: Move up valgrind and g-i checks. Make the python
13101 dependency optional, as it was before.
13103 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13105 Merge branch 'master' into 0.11
13110 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13112 * gst/rtsp-server/rtsp-media.c:
13113 media: update range when active clients changed
13114 When we changed the number of active clients, update the current range
13115 information because we want the second client connecting to a shared resource
13116 continue from where the stream currently.
13118 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13120 * gst/rtsp-server/rtsp-media-factory-uri.c:
13121 * gst/rtsp-server/rtsp-media-factory-uri.h:
13122 factory-uri: add colorspace and fix pt
13123 Rework the way we pass data to the autoplugger.
13124 When we have raw caps, plug a converter element to make pluggin to raw
13125 payloaders more successful.
13126 Make sure all dynamically plugged payloaders have a unique payload types.
13128 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13130 * examples/Makefile.am:
13131 * examples/test-uri.c:
13132 example: add example of the uri factory
13134 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13136 * gst/rtsp-server/Makefile.am:
13137 * gst/rtsp-server/rtsp-media-factory-uri.c:
13138 * gst/rtsp-server/rtsp-media-factory-uri.h:
13139 * gst/rtsp-server/rtsp-server.h:
13140 factory-uri: add a factory to stream any URI
13141 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
13144 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13146 * gst/rtsp-server/rtsp-media.c:
13147 * gst/rtsp-server/rtsp-media.h:
13148 media: ignore spurious ASYNC_DONE messages
13149 When we are dynamically adding pads, the addition of the udpsrc elements will
13150 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
13151 the real ASYNC_DONE when everything is prerolled.
13153 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13155 * gst/rtsp-server/rtsp-media-factory.c:
13156 * gst/rtsp-server/rtsp-media-factory.h:
13157 media-factory: make lock macro
13159 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
13161 * gst/rtsp-server/rtsp-client.c:
13162 rtsp-server: Remove unused variable and dead assignment
13164 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
13166 * examples/test-launch.c:
13167 * examples/test-mp4.c:
13168 * examples/test-ogg.c:
13169 * examples/test-readme.c:
13170 * examples/test-sdp.c:
13171 * examples/test-video.c:
13172 examples: Run gst-indent
13174 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
13176 * gst/rtsp-server/rtsp-client.c:
13177 * gst/rtsp-server/rtsp-media-factory.c:
13178 * gst/rtsp-server/rtsp-media-mapping.c:
13179 * gst/rtsp-server/rtsp-media.c:
13180 * gst/rtsp-server/rtsp-params.c:
13181 * gst/rtsp-server/rtsp-sdp.c:
13182 * gst/rtsp-server/rtsp-server.c:
13183 * gst/rtsp-server/rtsp-session-pool.c:
13184 * gst/rtsp-server/rtsp-session.c:
13185 rtsp-server: Run gst-indent
13186 Since it wasn't using the upstream common previously, there was no
13187 indentation check before commiting.
13189 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
13191 * gst/rtsp-server/rtsp-media-mapping.h:
13192 * gst/rtsp-server/rtsp-media.c:
13193 * gst/rtsp-server/rtsp-media.h:
13194 * gst/rtsp-server/rtsp-sdp.c:
13195 * gst/rtsp-server/rtsp-session-pool.h:
13196 * gst/rtsp-server/rtsp-session.c:
13197 * gst/rtsp-server/rtsp-session.h:
13198 rtsp-server: Some more doc fixups
13200 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13203 Makefile: Add cruft-cleaning support
13205 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13209 * docs/Makefile.am:
13210 * docs/libs/Makefile.am:
13211 * docs/libs/gst-rtsp-server-docs.sgml:
13212 * docs/libs/gst-rtsp-server-sections.txt:
13213 * docs/libs/gst-rtsp-server.types:
13214 * docs/version.entities.in:
13215 docs: Add gtk-doc build system
13217 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13219 * gst/rtsp-server/Makefile.am:
13220 Makefile.am: Use standard GIR make behaviour
13222 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
13226 autogen/configure: Bring more in sync to standard gst module behaviour
13228 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13230 * gst/rtsp-server/rtsp-media.c:
13231 media: warn and fail when gstrtpbin is not found
13233 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13236 configure: open 0.11 branch
13238 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
13242 Add common submodule
13244 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
13246 * common/ChangeLog:
13247 * common/Makefile.am:
13248 * common/c-to-xml.py:
13249 * common/check.mak:
13250 * common/coverage/coverage-report-entry.pl:
13251 * common/coverage/coverage-report.pl:
13252 * common/coverage/coverage-report.xsl:
13253 * common/coverage/lcov.mak:
13254 * common/gettext.patch:
13255 * common/glib-gen.mak:
13256 * common/gst-autogen.sh:
13257 * common/gst-xmlinspect.py:
13259 * common/gstdoc-scangobj:
13260 * common/gtk-doc-plugins.mak:
13261 * common/gtk-doc.mak:
13262 * common/m4/.gitignore:
13263 * common/m4/Makefile.am:
13264 * common/m4/README:
13265 * common/m4/as-ac-expand.m4:
13266 * common/m4/as-auto-alt.m4:
13267 * common/m4/as-compiler-flag.m4:
13268 * common/m4/as-compiler.m4:
13269 * common/m4/as-docbook.m4:
13270 * common/m4/as-libtool-tags.m4:
13271 * common/m4/as-libtool.m4:
13272 * common/m4/as-python.m4:
13273 * common/m4/as-scrub-include.m4:
13274 * common/m4/as-version.m4:
13275 * common/m4/ax_create_stdint_h.m4:
13276 * common/m4/check.m4:
13277 * common/m4/glib-gettext.m4:
13278 * common/m4/gst-arch.m4:
13279 * common/m4/gst-args.m4:
13280 * common/m4/gst-check.m4:
13281 * common/m4/gst-debuginfo.m4:
13282 * common/m4/gst-default.m4:
13283 * common/m4/gst-doc.m4:
13284 * common/m4/gst-error.m4:
13285 * common/m4/gst-feature.m4:
13286 * common/m4/gst-function.m4:
13287 * common/m4/gst-gettext.m4:
13288 * common/m4/gst-glib2.m4:
13289 * common/m4/gst-libxml2.m4:
13290 * common/m4/gst-plugindir.m4:
13291 * common/m4/gst-valgrind.m4:
13292 * common/m4/gtk-doc.m4:
13293 * common/m4/introspection.m4:
13294 * common/m4/pkg.m4:
13295 * common/mangle-tmpl.py:
13296 * common/plugins.xsl:
13298 * common/release.mak:
13299 * common/scangobj-merge.py:
13300 * common/upload.mak:
13301 common: Remove static version
13303 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
13305 * common/m4/introspection.m4:
13306 Update introspection.m4 to match usage
13308 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13312 Remove old stuff from the README
13314 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13317 back to development
13319 === release 0.10.7 ===
13321 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13326 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13328 * examples/test-ogg.c:
13329 test-ogg: remove parsers
13330 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
13331 buffers with timestamps. Using the parsers also seems to break things.
13333 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13335 * bindings/vala/gst-rtsp-server-0.10.vapi:
13336 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13337 Updated Vala bindings
13339 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13341 * common/m4/introspection.m4:
13343 * gst/rtsp-server/Makefile.am:
13344 Added initial gobject-introspection support
13346 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13348 * gst/rtsp-server/rtsp-media-factory.c:
13349 media-factory: don't use host for shared hash key
13350 When we generate the key to share made between connections, don't include the
13351 host used to connect so that we can share media even if between clients that
13352 connected with localhost and ones with the ip address.
13354 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13356 * bindings/vala/Makefile.am:
13357 build: fix distcheck
13359 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13361 * bindings/vala/gst-rtsp-server-0.10.vapi:
13362 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13363 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13364 Update Vala bindings
13366 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
13368 * bindings/vala/Makefile.am:
13370 Fix configure checks and installation location for Vala bindings
13373 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13376 back to development
13378 === release 0.10.6 ===
13380 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13383 configure: release 0.10.6
13385 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13387 * gst/rtsp-server/rtsp-media.c:
13388 media: help the compiler a little
13390 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13392 * gst/rtsp-server/rtsp-media.c:
13393 * gst/rtsp-server/rtsp-media.h:
13394 * gst/rtsp-server/rtsp-session.c:
13395 media: cleanup media transport before freeing
13396 Cleanup the media transport data before freeing. In particular, remove the qdata
13397 from the rtpsource object.
13399 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13401 * gst/rtsp-server/rtsp-media-factory.c:
13402 * gst/rtsp-server/rtsp-media-factory.h:
13403 * gst/rtsp-server/rtsp-media.c:
13404 * gst/rtsp-server/rtsp-media.h:
13405 media-factory: add eos-shutdown property
13406 Add an eos-shutdown property that will send an EOS to the pipeline before
13407 shutting it down. This allows for nice cleanup in case of a muxer.
13410 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13412 * gst/rtsp-server/rtsp-media.c:
13413 * gst/rtsp-server/rtsp-media.h:
13414 media: use multiudpsink send-duplicates when we can
13415 If we have a new enough multiudpsink with the send-duplicates property, use this
13416 instead of doing our own filtering. Our custom filtering code should eventually
13417 be removed when we can depend on a released -good.
13419 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13421 * gst/rtsp-server/rtsp-media.c:
13422 media: don't leak destinations
13423 Refactor and cleanup the destinations array when the stream is destroyed.
13425 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13427 * gst/rtsp-server/rtsp-media.c:
13428 * gst/rtsp-server/rtsp-media.h:
13429 media: don't add udp addresses multiple times
13430 Keep track of the udp addresses we added to udpsink and never add the same udp
13431 destination twice. This avoids duplicate packets when using multicast.
13433 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13435 * gst/rtsp-server/rtsp-server.c:
13436 server: disable use of SO_LINGER
13437 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
13438 server close()s the connection.
13440 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13442 * gst/rtsp-server/rtsp-server.c:
13443 server: use 5 second linger period in SO_LINGER
13444 Wait 5 seconds before clearing the send buffers and reseting the connection with
13445 the client when we do a close. This should be enough time to get the message to
13449 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
13451 * gst/rtsp-server/rtsp-server.c:
13452 server: use SO_LINGER
13453 SO_LINGER on the socket will make sure that any pending data on the socket is
13454 flushed ASAP and that the socket connection is reset. This makes sure that the
13455 socket can be reused immediately.
13458 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13461 README: add blurb about shared media factories
13463 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
13465 * gst/rtsp-server/rtsp-media.c:
13466 Add stdlib.h for atoi()
13468 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13470 * bindings/python/Makefile.am:
13471 * bindings/vala/Makefile.am:
13472 build: distcheck fixes
13473 Fix 'make distcheck', somewhat (it still fails because it tries to
13474 install files into /usr/share/vala/vapi/ irrespective of the
13475 configured prefix).
13477 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13480 configure: bump core/base requirements to released version
13481 Makes things less confusing for people.
13483 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13486 configure: fail if GStreamer core/base requirements are not met
13488 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13490 * gst/rtsp-server/rtsp-client.c:
13491 client: improve client cleanups
13492 Make sure the session does not timeout when using TCP. We need to do this
13493 because quicktime player does not send RTCP for some reason in tunneled
13495 Refactor some cleanup code.
13498 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13500 * gst/rtsp-server/rtsp-session.c:
13501 * gst/rtsp-server/rtsp-session.h:
13502 session: add support for prevent session timeouts
13503 Add an atomix counter to prevent session timeouts when we are, for example,
13504 streaming over TCP.
13506 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13508 * gst/rtsp-server/rtsp-client.c:
13509 client: fix unlink on session timeouts
13510 When our session times out, make sure we unlink all streams in this
13512 Remove the tunnelid when closing the connection.
13514 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13516 * gst/rtsp-server/rtsp-session.c:
13517 session: small cleanups
13519 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13521 * gst/rtsp-server/rtsp-client.c:
13522 client: handle lost_tunnel callbacks
13523 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
13524 hashtable so that we can reuse it for when the client reopens the POST
13526 Close the connection after a TEARDOWN.
13527 Make sure or watchid is cleared when the watch is removed.
13530 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13532 * gst/rtsp-server/rtsp-client.c:
13533 * gst/rtsp-server/rtsp-media.c:
13534 * gst/rtsp-server/rtsp-sdp.c:
13535 rtsp-server: add more support for multicast
13537 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13540 * gst/rtsp-server/rtsp-media.c:
13541 * gst/rtsp-server/rtsp-media.h:
13542 media: allow configuration of allowed lower transport
13544 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13546 * gst/rtsp-server/rtsp-client.h:
13547 * gst/rtsp-server/rtsp-media.c:
13548 * gst/rtsp-server/rtsp-media.h:
13549 * gst/rtsp-server/rtsp-sdp.c:
13550 * gst/rtsp-server/rtsp-sdp.h:
13551 * gst/rtsp-server/rtsp-server.c:
13552 rtsp: keep track of server ip and ipv6
13553 Keep track of how the client connected to the server and setup the udp ports
13554 with the same protocol.
13555 Copy the server ip address in the SDP so that clients can send RTCP back to
13558 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13560 * gst/rtsp-server/rtsp-session.c:
13563 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13565 * gst/rtsp-server/rtsp-client.c:
13566 client: use right size for malloc
13568 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13570 * gst/rtsp-server/rtsp-server.c:
13571 server: comment ipv6 server listening address
13573 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13575 * gst/rtsp-server/rtsp-media.c:
13576 media: allow for ipv6 sockets
13578 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13580 * gst/rtsp-server/rtsp-server.c:
13581 * gst/rtsp-server/rtsp-server.h:
13582 server: rework server part
13583 Allow setting a bind address, make sure we can deal with ipv6.
13584 Remove the port property and change with the service property.
13586 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13588 * gst/rtsp-server/rtsp-media.h:
13589 media: update comments a little
13591 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13593 * gst/rtsp-server/rtsp-client.c:
13594 client: make content-base better
13595 Use the URI formatting functions to make a content-base. Also make sure that
13596 there is a trailing / at the end.
13598 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13600 * gst/rtsp-server/rtsp-client.c:
13601 client: guard against invalid paths
13603 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13605 * examples/test-video.c:
13606 test: catch server bind errors
13608 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13610 * gst/rtsp-server/rtsp-media.c:
13611 rtspmedia: emit "unprepared" if _prepare fails.
13612 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13613 media object is removed from its factory's cache.
13615 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13617 * gst/rtsp-server/rtsp-media.c:
13618 media: collect media position when seek completes
13620 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13622 * gst/rtsp-server/rtsp-client.c:
13623 client: call unlink_streams in client finalize
13626 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13628 * gst/rtsp-server/rtsp-media.c:
13629 media: limit the time to wait to something huge
13630 Avoid waiting forever but limit the timeout to 20 seconds.
13632 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13634 * gst/rtsp-server/rtsp-sdp.c:
13635 sdp: reindent and check for prepared status
13637 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13639 * gst/rtsp-server/rtsp-media.c:
13640 * gst/rtsp-server/rtsp-media.h:
13641 * gst/rtsp-server/rtsp-session.c:
13642 media: avoid doing _get_state() for state changes
13643 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13644 until the media is prerolled or in error. This avoids doing a blocking call of
13645 gst_element_get_state() that can cause lockups when there is an error.
13648 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13650 * gst/rtsp-server/rtsp-media.c:
13653 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13655 * gst/rtsp-server/rtsp-media-factory.c:
13656 media-factory: better error handling
13657 Improve the error handling a bit.
13659 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13661 * gst/rtsp-server/rtsp-client.c:
13662 client: rework transport parsing
13663 Rework the transport parsing code so that we can ignore transports we don't
13664 support instead of just picking the first one we can parse.
13665 Configure a (for now hardcoded) destination for multicast transports.
13667 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13669 * gst/rtsp-server/rtsp-media.c:
13670 media: set multicast sink parameters
13671 Disable loop and automatic multicast join on the udpsink elements.
13672 Add some more debug info.
13673 Reset some state variables in the right place.
13674 Use the right port numbers for multicast.
13676 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13678 * gst/rtsp-server/rtsp-session.c:
13679 session: handle transport setup correctly
13680 Handle UDP, MCAST and TCP transport negotiation more correctly.
13681 Store the server session SSRC in the transport.
13683 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13685 * gst/rtsp-server/rtsp-client.c:
13686 rtsp-client: implement error_full
13687 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13690 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13693 * gst/rtsp-server/rtsp-client.c:
13694 * gst/rtsp-server/rtsp-server.c:
13695 docs: update docs and comments
13697 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13699 * gst/rtsp-server/rtsp-sdp.c:
13700 sdp: make server work better when behind a proxy
13702 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13704 * gst/rtsp-server/rtsp-client.c:
13705 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13707 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13709 * gst/rtsp-server/rtsp-client.c:
13710 * gst/rtsp-server/rtsp-media-factory.c:
13711 * gst/rtsp-server/rtsp-media-mapping.c:
13712 * gst/rtsp-server/rtsp-media.c:
13713 * gst/rtsp-server/rtsp-server.c:
13714 * gst/rtsp-server/rtsp-session-pool.c:
13715 * gst/rtsp-server/rtsp-session.c:
13716 Use GStreamer's debugging subsystem
13718 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13720 * gst/rtsp-server/rtsp-media-factory.c:
13721 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13723 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13726 back to development
13728 === release 0.10.5 ===
13730 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13735 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13738 configure: bump required versions
13740 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13742 * gst/rtsp-server/rtsp-client.c:
13743 client: call weak-unref on client->sessions from finalize
13746 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13748 * gst/rtsp-server/rtsp-media.c:
13749 media: Fixed crasher where caps got unref'ed too often
13751 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13754 * pkgconfig/.gitignore:
13755 * pkgconfig/Makefile.am:
13756 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13757 Added pkg-config file to use gst-rtsp-server uninstalled
13759 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13761 * gst/rtsp-server/rtsp-media.c:
13762 media: add some docs
13764 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13766 * gst/rtsp-server/rtsp-client.c:
13767 rtsp: Use gst_rtsp_watch_send_message().
13768 Use gst_rtsp_watch_send_message() since the old API which used
13769 gst_rtsp_watch_queue_message() has been deprecated.
13771 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13774 back to development
13776 === release 0.10.4 ===
13778 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13783 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13785 * gst/rtsp-server/rtsp-client.c:
13786 * gst/rtsp-server/rtsp-session.c:
13787 * gst/rtsp-server/rtsp-session.h:
13788 rtsp: allocate channels in TCP mode
13789 When the client does not provide us with channels in TCP mode, allocate channels
13792 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13794 * gst/rtsp-server/rtsp-client.c:
13795 client: don't crash when tunnelid is missing
13796 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13797 don't crash but return an error response to the client.
13800 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13802 * bindings/vala/gst-rtsp-server-0.10.vapi:
13803 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13804 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13805 bindings: update vala bindings with new method
13807 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13809 * gst/rtsp-server/rtsp-session-pool.c:
13810 * gst/rtsp-server/rtsp-session-pool.h:
13811 sessionpool: add function to filter sessions
13812 Add generic function to retrieve/remove sessions.
13814 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13817 configure: bump core/base requirements to release
13819 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13821 * gst/rtsp-server/rtsp-media.c:
13822 media: fix indentation
13824 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13826 * gst/rtsp-server/rtsp-media.c:
13827 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13829 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13831 * gst/rtsp-server/rtsp-media.c:
13832 set state and remove elements of media in for loop
13834 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13836 * bindings/vala/gst-rtsp-server-0.10.vapi:
13837 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13838 Added gst_rtsp_media_remove_elements function to Vala bindings
13840 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13842 * gst/rtsp-server/rtsp-media.c:
13843 * gst/rtsp-server/rtsp-media.h:
13844 Added gst_rtsp_media_remove_elements function
13846 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13848 * gst/rtsp-server/rtsp-media.c:
13849 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13851 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13853 * bindings/vala/gst-rtsp-server-0.10.vapi:
13854 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13855 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13856 Updated Vala bindings
13858 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13860 * gst/rtsp-server/rtsp-media.c:
13861 * gst/rtsp-server/rtsp-media.h:
13862 Added vmethod unprepare to GstRTSPMedia
13863 The default implementation sets the state of the pipeline to GST_STATE_NULL
13865 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13867 * gst/rtsp-server/rtsp-media-factory.c:
13868 * gst/rtsp-server/rtsp-media-factory.h:
13869 Made collect_streams function public
13871 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13873 * gst/rtsp-server/rtsp-media-factory.c:
13874 * gst/rtsp-server/rtsp-media-factory.h:
13875 * gst/rtsp-server/rtsp-media.c:
13876 Added vmethod create_pipeline to GstRTSPMediaFactory
13877 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13879 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13881 * gst/rtsp-server/rtsp-client.c:
13882 client: use g_source_destroy()
13883 We need to use g_source_destroy() because we might have added the source to a
13884 different main context than the default one.
13886 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13888 * gst/rtsp-server/Makefile.am:
13889 * gst/rtsp-server/rtsp-client.c:
13890 * gst/rtsp-server/rtsp-params.c:
13891 * gst/rtsp-server/rtsp-params.h:
13892 rtsp: prepare for handling GET/SET_PARAMETER
13893 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13895 Fix return codes of handlers.
13897 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13899 * gst/rtsp-server/rtsp-media.c:
13900 media: don't leak session pads
13902 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13904 * gst/rtsp-server/rtsp-media.c:
13905 media: clean up the messages a bit
13907 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13909 * gst/rtsp-server/rtsp-sdp.c:
13910 sdp: warn and skip streams without media
13912 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13914 * bindings/vala/gst-rtsp-server-0.10.vapi:
13915 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13916 vala: Fixed typo in header file of RTSPMediaStream
13918 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13920 * gst/rtsp-server/rtsp-media.c:
13922 Fix a debug message
13923 Make dumping RTCP stats configurable
13925 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13927 * gst/rtsp-server/rtsp-media.c:
13928 media: be less verbose and leak less
13930 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13932 * gst/rtsp-server/rtsp-media.c:
13933 media: don't leak the destination address
13935 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13937 * gst/rtsp-server/rtsp-client.c:
13938 * gst/rtsp-server/rtsp-media.c:
13939 * gst/rtsp-server/rtsp-media.h:
13940 * gst/rtsp-server/rtsp-session.c:
13941 * gst/rtsp-server/rtsp-session.h:
13942 rtsp: use RTCP to keep the session alive
13943 Use the RTCP rtcp-from stats field to find the associated session and use this
13944 to keep the session alive.
13946 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13948 * gst/rtsp-server/rtsp-session.c:
13949 session: add 5sec to the real session timeout
13950 Allow the session to live 5sec longer before really timing out. This should give
13951 clients some extra time to keep the session active.
13953 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13955 * gst/rtsp-server/rtsp-client.c:
13956 client: replay OK to GET/SET_PARAMETER
13957 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13958 so that we return OK for those requests.
13960 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13962 * gst/rtsp-server/rtsp-media.c:
13963 * gst/rtsp-server/rtsp-media.h:
13964 media: keep track of active transports
13965 Keep track of which transport is active to avoid closing the connection too
13967 Remove the destination transport also when going to NULL.
13968 Print some stats about the SDES and other RTCP messages we receive from the
13971 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13973 * examples/.gitignore:
13974 * examples/Makefile.am:
13975 * examples/test-sdp.c:
13976 example: add SDP relay example
13978 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13980 * gst/rtsp-server/rtsp-media.c:
13981 media: also count active TCP connections
13983 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13985 * gst/rtsp-server/rtsp-media-factory.c:
13986 * gst/rtsp-server/rtsp-media.c:
13987 * gst/rtsp-server/rtsp-media.h:
13988 rtsp: add support for dynamic elements
13989 Add support for dynamic elements.
13990 Don't set live pipelines back to paused.
13992 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13994 * gst/rtsp-server/rtsp-sdp.c:
13995 sdp: don't add encoding name when absent in caps
13997 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13999 * gst/rtsp-server/rtsp-client.c:
14000 client: warn when we can't do RTP-Info
14002 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14004 * gst/rtsp-server/rtsp-media-factory.c:
14005 factory: factor out the stream construction
14007 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14009 * gst/rtsp-server/rtsp-client.c:
14010 client: only add RTP-Info when we have the info
14011 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
14014 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14017 back to development
14019 === release 0.10.3 ===
14021 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14025 - Fixes a bug where it put the wrong verion in pkgconfig
14026 - Link RTP and RTCP sources
14028 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14030 * gst/rtsp-server/rtsp-media.c:
14031 * gst/rtsp-server/rtsp-media.h:
14032 media: link the RTP udpsrc to the session manager
14033 Link the RTP udpsrc and the appsrc to the session manager so that they don't
14034 shut down when the client sends a packet to open firewalls.
14036 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14038 * pkgconfig/gst-rtsp-server.pc.in:
14039 Don't use hard-coded version number in pkg-config file
14041 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14044 back to development
14046 === release 0.10.2 ===
14048 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14053 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14056 * common/m4/.gitignore:
14057 * examples/.gitignore:
14058 * pkgconfig/.gitignore:
14059 add some .gitignore files
14061 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14063 * gst/rtsp-server/rtsp-media.c:
14064 media: seek to key frames
14066 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14068 * gst/rtsp-server/rtsp-media.c:
14069 media: emit the unprepared signal by id
14070 Emit the unprepared signal by id instead of name and set the media as
14073 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14075 * gst/rtsp-server/rtsp-media.c:
14076 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
14078 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14080 * gst/rtsp-server/rtsp-server.c:
14081 Added finalize function to GstRTPSPServer to unref session pool and media mapping
14083 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
14085 * bindings/vala/gst-rtsp-server-0.10.vapi:
14086 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14087 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14088 Updated vala bindings
14090 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14092 * gst/rtsp-server/Makefile.am:
14093 * gst/rtsp-server/rtsp-client.c:
14094 * gst/rtsp-server/rtsp-media.c:
14095 server: use appsink and appsrc with the API
14096 Use the appsink/appsrc API instead of the signals for higher
14099 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14101 * examples/test-ogg.c:
14102 tests: set the payload type correctly
14104 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14106 * gst/rtsp-server/rtsp-media-factory.c:
14107 factory: connect to the unprepare signal
14108 Connect to the unprepare signal for non-reusable media so that we can remove
14109 them from the cache.
14111 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14113 * gst/rtsp-server/rtsp-media.c:
14114 * gst/rtsp-server/rtsp-media.h:
14115 media: add signal to notify of unprepare
14117 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14119 * gst/rtsp-server/rtsp-media.c:
14120 * gst/rtsp-server/rtsp-media.h:
14121 media: more work on making the media shared
14122 Add a reusable flag to medias, indicating that they can be reused after a state
14126 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14128 * examples/test-readme.c:
14129 examples: mark the example as shared for testing
14131 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14133 * gst/rtsp-server/rtsp-media.c:
14134 * gst/rtsp-server/rtsp-media.h:
14135 client: support shared media
14136 Always perform the state actions even if the target state of the pipeline is
14137 already correct, we still want to add/remove the transports when we are dealing
14139 Keep a counter of the number of active transports for a media so that we can use
14140 this to perform a state change when needed.
14141 Perform a state change of the pipeline only when the first transport was added
14142 or when there are no active transports.
14144 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
14146 * gst/rtsp-server/rtsp-client.c:
14147 client: fix refcounting crasher
14148 Don't need to remove the weak refs in the finalize methods, they are already
14149 removed in the dispose.
14150 Don't register the callback with a DestroyNofity.
14152 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14154 * gst/rtsp-server/rtsp-client.c:
14155 Fix rtsp client refcount management in TCP mode.
14156 Don't unref a client ref we never had. Fixes an unref
14157 of an already-free client object after a client
14158 teardown request for me.
14160 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
14162 * gst/rtsp-server/rtsp-session.c:
14163 docs: fix typo in API docs
14165 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14167 * gst/rtsp-server/rtsp-media.c:
14168 More seeking fixes.
14169 Keep the udp sources in playing even if we go to paused. unlock the sources when
14171 Add some more debug info.
14172 Only seek when we need to.
14173 Keep track of the position when we go to paused.
14175 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14177 * gst/rtsp-server/rtsp-client.c:
14178 * gst/rtsp-server/rtsp-media.c:
14179 * gst/rtsp-server/rtsp-media.h:
14180 Add beginnings of seeking.
14181 Parse the Range header and perform a seek on the pipeline for the requested
14182 position. It's disabled currently until I figure out what's going wrong.
14184 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14186 * gst/rtsp-server/rtsp-client.c:
14187 allow pause requests for now.
14190 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14192 * gst/rtsp-server/rtsp-client.c:
14193 Remove weak ref on the session in teardown
14194 We need to remove our weakref from the session when we do a teardown because
14195 else we close the TCP connection prematurely.
14197 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14199 * gst/rtsp-server/rtsp-client.c:
14200 * gst/rtsp-server/rtsp-client.h:
14201 * gst/rtsp-server/rtsp-session-pool.c:
14202 Do some more session cleanup
14203 Make session timeout kill the TCP connection that currently watches the
14205 Remove the client timeout property.
14207 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14209 * gst/rtsp-server/rtsp-client.c:
14210 * gst/rtsp-server/rtsp-client.h:
14211 * gst/rtsp-server/rtsp-media.c:
14212 * gst/rtsp-server/rtsp-media.h:
14213 * gst/rtsp-server/rtsp-server.c:
14214 * gst/rtsp-server/rtsp-session.c:
14215 * gst/rtsp-server/rtsp-session.h:
14217 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
14220 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14222 * examples/Makefile.am:
14223 * examples/test-launch.c:
14224 Add example server that takes launch lines
14225 Add an example server that streams any -launch line.
14227 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14229 * examples/test-readme.c:
14230 * gst/rtsp-server/rtsp-client.c:
14231 * gst/rtsp-server/rtsp-media.c:
14232 * gst/rtsp-server/rtsp-media.h:
14233 Add support for live streams
14234 Add support for live streams and ranges
14235 Start on handling TCP data transfer.
14237 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14239 * gst/rtsp-server/rtsp-media.c:
14240 Free the pipeline before other things
14243 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14245 * gst/rtsp-server/rtsp-client.c:
14246 Only free the pending tunnel if there is one
14249 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14251 * gst/rtsp-server/rtsp-client.c:
14252 * gst/rtsp-server/rtsp-client.h:
14253 * gst/rtsp-server/rtsp-media.c:
14254 rtsp-server: Add support for tunneling
14255 Add support for tunneling over HTTP.
14256 Use new connection methods to retrieve the url.
14257 Dispatch messages based on the message type instead of blindly
14258 assuming it's always a request.
14259 Keep track of the watch id so that we can remove it later.
14260 Set the media pipeline to NULL before unreffing the pipeline.
14262 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14264 * gst/rtsp-server/rtsp-client.c:
14265 * gst/rtsp-server/rtsp-client.h:
14266 Fix for channel -> watch rename in gstreamer
14267 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
14269 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14271 * gst/rtsp-server/rtsp-client.c:
14272 * gst/rtsp-server/rtsp-client.h:
14274 Use the async RTSP channels instead of spawning a new thread for each client.
14275 If a sessionid is specified in a request, fail if we don't have the session.
14277 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14279 * gst/rtsp-server/rtsp-media.c:
14280 Add better debug info
14281 Add some better debug info.
14283 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14285 * examples/test-video.c:
14287 Add support for session timeouts in the example.
14289 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14291 * gst/rtsp-server/rtsp-session-pool.c:
14292 * gst/rtsp-server/rtsp-session-pool.h:
14293 Pass GTimeVal around for performance reasons
14294 Get the current time only once and pass it around so that sessions don't have to
14295 get the current time anymore.
14296 Add experimental support for a GSource that dispatches when the session needs to
14299 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14301 * gst/rtsp-server/rtsp-session.c:
14302 * gst/rtsp-server/rtsp-session.h:
14303 Add better support for session timeouts
14304 Add a method to request the number of milliseconds when a session will timeout.
14306 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14308 * gst/rtsp-server/rtsp-media.c:
14309 * gst/rtsp-server/rtsp-media.h:
14310 Add suport for RTP manager monitoring
14311 Add the first stage in monitoring the rtp manager.
14312 Make sure we don't update the state to something we don't want.
14314 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14316 * gst/rtsp-server/rtsp-client.c:
14317 Add support for session keepalive
14318 Get and update the session timeout for all requests. get the session as early as
14321 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14323 * gst/rtsp-server/rtsp-media-factory.h:
14324 * gst/rtsp-server/rtsp-media.c:
14325 * gst/rtsp-server/rtsp-media.h:
14326 Handle media bus messages
14327 Handle media bus messages in a custom mainloop and dispatch them to the
14328 RTSPMedia objects. Let the default implementation handle some common messages.
14330 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14332 * gst/rtsp-server/rtsp-client.c:
14333 * gst/rtsp-server/rtsp-session-pool.c:
14334 * gst/rtsp-server/rtsp-session.c:
14335 Some more session timeout handling
14336 Move the session header setting code to a central place so that we always add
14337 the timeout parameter too.
14338 Handle timeouts by running the session cleanup code.
14339 Stop media before cleaning up.
14341 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14343 * gst/rtsp-server/rtsp-client.c:
14344 * gst/rtsp-server/rtsp-client.h:
14345 Add timeout property
14346 Add a timeout property ot the client and make the other properties into GObject
14349 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14351 * gst/rtsp-server/rtsp-session-pool.c:
14352 Use getters and setters in property code
14353 Use the getters and setters for the timeout property instead of locking
14356 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14358 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
14360 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14362 * gst/rtsp-server/rtsp-session-pool.c:
14363 * gst/rtsp-server/rtsp-session-pool.h:
14364 * gst/rtsp-server/rtsp-session.c:
14365 * gst/rtsp-server/rtsp-session.h:
14366 Add more timeout stuff
14367 Add method to check if a session is expired.
14368 Add method to perform cleanup on a session pool.
14370 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14372 * gst/rtsp-server/rtsp-client.c:
14373 * gst/rtsp-server/rtsp-session-pool.c:
14374 * gst/rtsp-server/rtsp-session-pool.h:
14375 * gst/rtsp-server/rtsp-session.c:
14376 * gst/rtsp-server/rtsp-session.h:
14377 Add beginnings of session timeouts and limits
14378 Add the timeout value to the Session header for unusual timeout values.
14379 Allow us to configure a limit to the amount of active sessions in a pool. Set a
14380 limit on the amount of retry we do after a sessionid collision.
14381 Add properties to the sessionid and the timeout of a session. Keep track of
14382 creation time and last access time for sessions.
14384 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14386 * gst/rtsp-server/rtsp-client.c:
14387 * gst/rtsp-server/rtsp-media.c:
14388 * gst/rtsp-server/rtsp-media.h:
14389 * gst/rtsp-server/rtsp-sdp.c:
14390 * gst/rtsp-server/rtsp-session-pool.c:
14391 * gst/rtsp-server/rtsp-session.c:
14392 * gst/rtsp-server/rtsp-session.h:
14393 Cleanup of sessions and more
14394 Fix the refcounting of media and sessions in the client. Properly clean up the
14395 session data when the client performs a teardown.
14396 Add Server header to responses.
14397 Allow for multiple uri setups in one session.
14398 Add Range header to the PLAY response and add the range attribute to the SDP
14400 Fix the session pool remove method, it used the wrong key in the hashtable. Also
14401 give the ownership of the sessionid to the session object.
14403 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14405 * gst/rtsp-server/rtsp-server.c:
14406 * gst/rtsp-server/rtsp-server.h:
14408 Rename the 'server_port' variable to simply 'port'.
14410 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14413 * gst/rtsp-server/rtsp-client.c:
14414 * gst/rtsp-server/rtsp-media.c:
14415 * gst/rtsp-server/rtsp-media.h:
14416 * gst/rtsp-server/rtsp-session.c:
14417 * gst/rtsp-server/rtsp-session.h:
14418 Rework the way we handle transports for streams
14419 Make the media accept an array of transports for the streams that we have
14420 configured for the play/pause requests.
14421 Implement server states for a client and its media.
14422 Require 0.10.22.1 (git HEAD) of gstreamer.
14424 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14426 * gst/rtsp-server/rtsp-client.c:
14427 * gst/rtsp-server/rtsp-media-factory.c:
14428 Drop const from functions dealing with urls
14429 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
14430 have the right const in them.
14432 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14434 * gst/rtsp-server/rtsp-client.c:
14435 * gst/rtsp-server/rtsp-media.c:
14436 * gst/rtsp-server/rtsp-sdp.c:
14440 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14442 * gst/rtsp-server/rtsp-client.c:
14443 * gst/rtsp-server/rtsp-media-factory.c:
14444 * gst/rtsp-server/rtsp-media.c:
14445 * gst/rtsp-server/rtsp-media.h:
14447 Don't keep a reference to the GstRTSPMedia in the stream.
14448 Free more things when freeing the GstRTSPMedia.
14450 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14453 * gst/rtsp-server/rtsp-media-factory.c:
14454 * gst/rtsp-server/rtsp-media-factory.h:
14455 * gst/rtsp-server/rtsp-media.c:
14456 * gst/rtsp-server/rtsp-media.h:
14457 * gst/rtsp-server/rtsp-server.c:
14458 * gst/rtsp-server/rtsp-server.h:
14459 More docs and small cleanups
14460 Add some more docs and update the README
14461 Cleanup some method names.
14462 Remove an unneeded idx field in the GstRTSPMediaStream
14464 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14467 * examples/Makefile.am:
14468 * examples/test-readme.c:
14469 Add a README and more example code
14470 Add a README file that contains a small introduction on how to use the server
14471 along with the example code explained in the readme.
14473 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14475 * gst/rtsp-server/rtsp-media.c:
14476 * gst/rtsp-server/rtsp-server.c:
14477 Fix some leaks and change default port
14478 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
14479 we finished the initial preroll. If we keep them locked, setting the pipeline to
14480 NULL will not stop and clean up the sources correctly.
14481 Change the default RTSP port to 8554 aka the official alternative RTSP port.
14483 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14485 * gst/rtsp-server/rtsp-session.c:
14486 * gst/rtsp-server/rtsp-session.h:
14487 Cleanups to the session object
14488 Remove some unneeded variables in the session state of a stream such as the
14489 owner media and the server transport.
14490 Get the configuration of a media stream in a session based on the media_stream
14491 in the original object instead of our cached index.
14492 Free more data in the finalize method.
14494 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14496 * gst/rtsp-server/rtsp-client.c:
14497 * gst/rtsp-server/rtsp-client.h:
14498 Cleanups and reuse media from DESCRIBE
14499 Handle thread create errors.
14500 Rename some internal methods to better match what they actually do.
14501 Handle misconfiguration of session_pool and media_mapping gracefully.
14502 Cache the DESCRIBE media and uri in the client connection and reuse them when
14503 we receive a SETUP request in the same connection for the same uri.
14504 Cleanup the client connection object.
14506 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14508 * gst/rtsp-server/rtsp-media-factory.c:
14509 * gst/rtsp-server/rtsp-media-factory.h:
14510 * gst/rtsp-server/rtsp-media.c:
14511 * gst/rtsp-server/rtsp-media.h:
14512 Add shared properties to media and factory
14513 Add the shared property to media.
14514 Implement some simple caching in the factory depending on if the media is shared
14517 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14519 * gst/rtsp-server/rtsp-client.c:
14520 Add a little comment
14521 Add some comment about the content-base header.
14523 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14525 * examples/Makefile.am:
14526 * examples/test-mp4.c:
14527 * examples/test-ogg.c:
14528 * examples/test-video.c:
14529 * gst/rtsp-server/Makefile.am:
14530 * gst/rtsp-server/rtsp-client.c:
14531 * gst/rtsp-server/rtsp-client.h:
14532 * gst/rtsp-server/rtsp-media-factory.c:
14533 * gst/rtsp-server/rtsp-media-factory.h:
14534 * gst/rtsp-server/rtsp-media.c:
14535 * gst/rtsp-server/rtsp-media.h:
14536 * gst/rtsp-server/rtsp-sdp.c:
14537 * gst/rtsp-server/rtsp-sdp.h:
14538 * gst/rtsp-server/rtsp-server.c:
14539 * gst/rtsp-server/rtsp-server.h:
14540 * gst/rtsp-server/rtsp-session.c:
14541 * gst/rtsp-server/rtsp-session.h:
14542 Reorganize things, prepare for media sharing
14543 Added various other test server examples
14544 Move the SDP message generation to a separate helper.
14545 Refactor common code for finding the session.
14546 Add content-base for realplayer compatibility
14547 Clean up request uris before processing for better vlc compatibility.
14548 Move prerolling and pipeline construction to the RTSPMedia object.
14549 Use multiudpsink for future pipeline reuse.
14551 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14554 Back to development
14557 === release 0.10.1 ===
14559 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14562 Make 0.10.1 release
14565 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14567 * bindings/vala/Makefile.am:
14569 Add more directories and files to the dist.
14571 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14573 * bindings/python/Makefile.am:
14574 * bindings/python/rtspserver.override:
14575 Fixed compile error of python bindings
14577 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14579 * bindings/vala/gst-rtsp-server-0.10.vapi:
14580 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14581 Marked values as nullable accordingly
14583 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14585 * bindings/vala/gst-rtsp-server-0.10.vapi:
14586 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14587 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14588 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14589 Updated Vala bindings
14591 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14593 * gst/rtsp-server/rtsp-client.c:
14594 * gst/rtsp-server/rtsp-media-mapping.c:
14595 * gst/rtsp-server/rtsp-media-mapping.h:
14596 * gst/rtsp-server/rtsp-media.h:
14597 * gst/rtsp-server/rtsp-session-pool.h:
14598 Cleanups and doc updates
14599 Add some more documentation and do some minor cleanups here and there.
14601 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14603 * gst/rtsp-server/rtsp-client.c:
14604 * gst/rtsp-server/rtsp-media-factory.c:
14605 * gst/rtsp-server/rtsp-media-factory.h:
14606 * gst/rtsp-server/rtsp-media.c:
14607 * gst/rtsp-server/rtsp-media.h:
14608 * gst/rtsp-server/rtsp-session.c:
14609 * gst/rtsp-server/rtsp-session.h:
14611 Rename GstRTSPMediaBin to GstRTSPMedia
14612 Parse the request url into a GstRTSPUri object and pass this object to the
14613 various handlers and methods that require the uri.
14615 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14619 Add some more docs and remove some old code from the example.
14621 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14623 * gst/rtsp-server/rtsp-client.c:
14624 Handle state change failures better
14625 Handle state change failures better when changing the state of the pipeline to
14628 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14630 * gst/rtsp-server/rtsp-media-factory.c:
14631 * gst/rtsp-server/rtsp-media-factory.h:
14632 Make element creation more extendible
14633 Add get_element vmethod to the default MediaFactory so that subclasses can just
14634 override that method and still use the default logic for making a MediaBin from
14637 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14640 * gst/rtsp-server/Makefile.am:
14641 * gst/rtsp-server/rtsp-client.c:
14642 * gst/rtsp-server/rtsp-client.h:
14643 * gst/rtsp-server/rtsp-media-factory.c:
14644 * gst/rtsp-server/rtsp-media-factory.h:
14645 * gst/rtsp-server/rtsp-media-mapping.c:
14646 * gst/rtsp-server/rtsp-media-mapping.h:
14647 * gst/rtsp-server/rtsp-media.c:
14648 * gst/rtsp-server/rtsp-media.h:
14649 * gst/rtsp-server/rtsp-server.c:
14650 * gst/rtsp-server/rtsp-server.h:
14651 * gst/rtsp-server/rtsp-session.c:
14652 * gst/rtsp-server/rtsp-session.h:
14653 Make the server handle arbitrary pipelines
14654 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14655 The GstMediaBin object has a handle to a bin with elements and to a list of
14656 GstMediaStream objects that this bin produces.
14657 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14658 with methods to register and remove those mappings.
14659 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14660 used by the server instance.
14661 Modify the example application so that it shows how to create custom pipelines
14662 attached to a specific mount point.
14663 Various misc cleanps.
14665 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14667 * gst/rtsp-server/rtsp-server.c:
14668 * gst/rtsp-server/rtsp-server.h:
14669 Allow setting a custom media factory for a server
14671 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14673 * gst/rtsp-server/rtsp-client.c:
14674 * gst/rtsp-server/rtsp-client.h:
14675 Allow setting a custom media factory for a client.
14677 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14679 * gst/rtsp-server/Makefile.am:
14680 Add Makefile entry for the media factory
14682 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14684 * gst/rtsp-server/rtsp-media-factory.c:
14685 * gst/rtsp-server/rtsp-media-factory.h:
14686 Add media factory to map urls to media pipeline objects.
14688 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14690 * gst/rtsp-server/rtsp-media.c:
14691 * gst/rtsp-server/rtsp-media.h:
14692 Add comments. Remove unused field
14694 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14696 * gst/rtsp-server/rtsp-session-pool.c:
14697 * gst/rtsp-server/rtsp-session-pool.h:
14698 Allow custom session pools to override the session id allocation algorithms Add some comments.
14700 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14702 * gst/rtsp-server/rtsp-session.h:
14705 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14707 * gst/rtsp-server/rtsp-client.c:
14708 * gst/rtsp-server/rtsp-client.h:
14709 Move the connection code in one place Add some comments
14711 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14713 * gst/rtsp-server/rtsp-server.c:
14714 * gst/rtsp-server/rtsp-server.h:
14715 Make vmethod to create and accept new clients. Add some docs.
14717 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14719 * gst/rtsp-server/rtsp-server.c:
14720 * gst/rtsp-server/rtsp-server.h:
14721 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14723 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14725 * gst/rtsp-server/rtsp-client.c:
14726 * gst/rtsp-server/rtsp-client.h:
14727 Name the parameters more appropriately.
14729 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14731 * gst/rtsp-server/rtsp-session-pool.c:
14732 Do some more cleanup of the session pool.
14734 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14736 * gst/rtsp-server/Makefile.am:
14737 * gst/rtsp-server/rtsp-client.c:
14738 Check if return value of gst_rtsp_session_get_media is not NULL
14740 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14742 * gst/rtsp-server/Makefile.am:
14743 Install rtsp-session and rtsp-session-pool headers
14745 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14750 * bindings/python/Makefile.am:
14751 * bindings/python/arg-types.py:
14752 * bindings/python/codegen/Makefile.am:
14753 * bindings/python/codegen/__init__.py:
14754 * bindings/python/codegen/argtypes.py:
14755 * bindings/python/codegen/code-coverage.py:
14756 * bindings/python/codegen/codegen.py:
14757 * bindings/python/codegen/definitions.py:
14758 * bindings/python/codegen/defsparser.py:
14759 * bindings/python/codegen/docextract.py:
14760 * bindings/python/codegen/docgen.py:
14761 * bindings/python/codegen/fileprefix.override:
14762 * bindings/python/codegen/fileprefixmodule.c:
14763 * bindings/python/codegen/h2def.py:
14764 * bindings/python/codegen/mergedefs.py:
14765 * bindings/python/codegen/mkskel.py:
14766 * bindings/python/codegen/override.py:
14767 * bindings/python/codegen/reversewrapper.py:
14768 * bindings/python/codegen/scmexpr.py:
14769 * bindings/python/rtspserver-types.defs:
14770 * bindings/python/rtspserver.defs:
14771 * bindings/python/rtspserver.override:
14772 * bindings/python/rtspservermodule.c:
14774 Add python bindings.
14776 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14778 * bindings/Makefile.am:
14780 Don't go into python dir when requirements for python bindings are missing
14782 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14784 * bindings/Makefile.am:
14785 * bindings/vala/Makefile.am:
14787 Install Vala bindings if vala is available
14789 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14791 * bindings/vala/gst-rtsp-server-0.10.deps:
14792 * bindings/vala/gst-rtsp-server-0.10.vapi:
14793 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14794 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14795 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14796 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14797 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14798 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14799 Regenerated Vala bindings
14801 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14803 * bindings/vala/gst-rtsp-server.vapi:
14804 * bindings/vala/packages/gst-rtsp-server.metadata:
14805 Fixed typo in included headers for vala bindings
14807 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14811 * pkgconfig/Makefile.am:
14812 * pkgconfig/gst-rtsp-server.pc.in:
14813 Added pkgconfig file
14815 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14817 * bindings/vala/gst-rtsp-server.vapi:
14818 * bindings/vala/packages/gst-rtsp-server.excludes:
14819 * bindings/vala/packages/gst-rtsp-server.gi:
14820 * bindings/vala/packages/gst-rtsp-server.metadata:
14821 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14823 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14825 * bindings/vala/gst-rtsp-server.vapi:
14826 * bindings/vala/packages/gst-rtsp-server.deps:
14827 * bindings/vala/packages/gst-rtsp-server.files:
14828 * bindings/vala/packages/gst-rtsp-server.gi:
14829 * bindings/vala/packages/gst-rtsp-server.metadata:
14830 * bindings/vala/packages/gst-rtsp-server.namespace:
14831 Added Vala bindings
14833 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14835 * gst/rtsp-server/rtsp-session.c:
14836 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14838 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14840 * examples/Makefile.am:
14841 * gst/rtsp-server/Makefile.am:
14842 Put GStreamer version in library name
14844 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14846 * examples/Makefile.am:
14847 * gst/rtsp-server/Makefile.am:
14848 Fix some issues to pass distcheck
14850 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14852 * gst/rtsp-server/rtsp-server.c:
14853 Added port property to GstRTSPServer class.
14855 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14860 * examples/Makefile.am:
14863 * gst/rtsp-server/Makefile.am:
14864 * gst/rtsp-server/rtsp-client.c:
14865 * gst/rtsp-server/rtsp-client.h:
14866 * gst/rtsp-server/rtsp-media.c:
14867 * gst/rtsp-server/rtsp-media.h:
14868 * gst/rtsp-server/rtsp-server.c:
14869 * gst/rtsp-server/rtsp-server.h:
14870 * gst/rtsp-server/rtsp-session-pool.c:
14871 * gst/rtsp-server/rtsp-session-pool.h:
14872 * gst/rtsp-server/rtsp-session.c:
14873 * gst/rtsp-server/rtsp-session.h:
14875 Split in library and example program
14877 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14879 * src/rtsp-client.h:
14880 Removed obsolete variable
14882 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14884 * src/rtsp-client.c:
14885 * src/rtsp-client.h:
14886 Removed pipeline variable GstRTSPClient, because it's only used in one function
14888 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14890 * src/rtsp-media.c:
14891 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14893 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14895 * src/rtsp-session.c:
14896 Initialize some more vars.
14898 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14900 * src/rtsp-session.c:
14901 Initialize variable to avoid compiler warning.
14903 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14906 Add a reasonable generic .gitignore