3 2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
8 * gst-rtsp-server.doap:
12 2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
14 * gst/rtsp-server/rtsp-media.c:
15 * gst/rtsp-server/rtsp-stream.c:
16 * gst/rtsp-server/rtsp-stream.h:
17 * gst/rtsp-sink/gstrtspclientsink.c:
18 Protection against early RTCP packets.
19 When receiving RTCP packets early the funnel is not ready yet and
20 GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
21 This causes the thread that handle RTCP packets to go to pause mode.
22 Since this thread is in pause mode there will be no further callbacks to
23 handle keep-alive for incoming RTCP packets. This will make the session
24 time out if the client is not using another keep-alive mechanism.
25 Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
26 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
28 2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
32 Update COPYING.LIB, COPYING files
33 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
35 2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
37 * docs/gst_plugins_cache.json:
41 === release 1.19.1 ===
43 2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
48 * docs/gst_plugins_cache.json:
49 * gst-rtsp-server.doap:
53 2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
55 * gst/rtsp-server/rtsp-stream.c:
56 rtsp-stream: use new gst_buffer_new_memdup()
57 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
59 2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
61 * gst/rtsp-server/rtsp-media-factory-uri.c:
62 rtsp-media: fix leak when adding converter
63 Free the previous caps before reusing the variable for the converter caps.
64 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
66 2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
68 * gst/rtsp-server/rtsp-client.c:
69 rtsp-client: fix leak adding headers
70 gst_rtsp_message_add_header() makes a copy of the header, instead
72 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
74 2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
76 * gst/rtsp-server/rtsp-stream.c:
77 Use gst_element_request_pad_simple...
78 Instead of the deprecated gst_element_get_request_pad.
79 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
81 2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
83 * gst/rtsp-server/rtsp-media.c:
84 rtsp-media: Ensure the bus watch is removed during unprepare
85 It's possible for the destruction of the source to be delayed.
86 Instead of relying on the dispose() to remove the bus watch, do
88 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
90 2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
93 docs: minor spelling correction in README
94 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
96 2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
98 * examples/test-replay-server.c:
99 test-replay-server: minor spelling corrections
100 Bumped on these while investigating the example code.
101 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
103 2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
105 * tests/check/gst/stream.c:
106 tests: Don't fail tests if IPv6 not available.
107 On computers with IPv6 disabled it shouldn't result in a test failure.
108 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
110 2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
112 * gst/rtsp-server/rtsp-media.c:
113 rtsp-media: Add one more case to seek avoidance
114 This is an extension to the previous commit. There can also be cases where the
115 start position is not specified, in those cases we should also avoid doing
116 seeking unless it's forced.
117 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
119 2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
121 * gst/rtsp-server/rtsp-media.c:
122 rtsp-media: Improve skipping trickmode seek.
123 We can also skip the seek if the end range is already
125 Avoids initial seek on play start if playing full stream.
126 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
128 2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
130 * gst/rtsp-sink/gstrtspclientsink.c:
131 rtspclientsink: Don't run signal class handlers during the CLEANUP stage
132 It's sufficient to run them during the FIRST stage instead of in both.
133 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
135 2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
137 * tests/check/gst/rtspclientsink.c:
138 tests: rtspclientsink: fix some leaks
139 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
141 2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
143 * gst/rtsp-sink/gstrtspclientsink.c:
144 rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
145 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
147 2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
149 * tests/check/gst/rtspclientsink.c:
150 rtspclientsink: add unit test for potential shutdown deadlock
151 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
153 2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
155 * gst/rtsp-sink/gstrtspclientsink.c:
156 rtspclientsink: fix deadlock on shutdown before preroll
157 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
158 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
160 2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
162 * gst/rtsp-server/rtsp-stream.c:
163 rtsp-stream: avoid deadlock in send_func
164 Currently the send_func() runs in a thread of its own which is started
165 the first time we enter handle_new_sample(). It runs in an outer loop
166 until priv->continue_sending is FALSE, which happens when a TEARDOWN
167 request is received. We use a local variable, cont, which is initialized
168 to TRUE, meaning that we will always enter the outer loop, and at the
169 end of the outer loop we assign it the value of priv->continue_sending.
170 Within the outer loop there is an inner loop, where we wait to be
171 signaled when there is more data to send. The inner loop is exited when
172 priv->send_cookie has changed value, which it does when more data is
173 available or when a TEARDOWN has been received.
174 But if we get a TEARDOWN before send_func() is entered we will get stuck
175 in the inner loop because no one will increase priv->session_cookie
177 By not entering the outer loop in send_func() if priv->continue_sending
178 is FALSE we make sure that we do not get stuck in send_func()'s inner
179 loop should we receive a TEARDOWN before the send thread has started.
180 Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
181 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
183 2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
185 * gst/rtsp-server/rtsp-client.c:
186 rtsp-client: cleanup transports during TEARDOWN
187 When tunneling RTP over RTSP the stream transports are stored in a hash
188 table in the GstRTSPClientPrivate struct. They are used for, among other
189 things, mapping channel id to stream transports when receiving data from
190 the client. The stream tranports are created and added to the hash table
191 in handle_setup_request(), but unfortuately they are not removed in
192 handle_teardown_request(). This means that if the client sends data on
193 the RTSP connection after it has sent the TEARDOWN, which is often the
194 case when audio backchannel is enabled, handle_data() will still be able
195 to map the channel to a session transport and pass the data along to it.
196 Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
197 because the stream is no longer joined to a bin.
198 We avoid this by removing the stream transports from the hash table when
199 we handle the TEARDOWN request.
200 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
202 2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
204 * docs/gst_plugins_cache.json:
205 * gst/rtsp-sink/gstrtspclientsink.c:
206 rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
209 2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
211 * tests/check/gst/client.c:
212 Add test cases for mountpoint of '/'
213 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
215 2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
217 * gst/rtsp-server/rtsp-client.c:
218 * gst/rtsp-server/rtsp-mount-points.c:
219 * gst/rtsp-server/rtsp-session-media.c:
220 Make a mount point of "/" work correctly.
221 As far as I can tell, this is neither explicitly allowed nor
222 forbidden by RFC 7826.
223 Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
224 use in the wild (presumably with non-GStreamer servers).
225 GStreamer's prior behavior was confusing, in that
226 gst_rtsp_mount_points_add_factory() would appear to accept a mount
227 path of "" or "/", but later connection attempts would fail with a
228 "media not found" error.
229 This commit makes a mount path of "/" work for either form of URL,
230 while an empty mount path ("") is rejected and logs a warning.
231 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
233 2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
235 * docs/gst_plugins_cache.json:
236 * gst/rtsp-sink/gstrtspclientsink.c:
237 rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
238 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
240 2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
242 * gst/rtsp-server/rtsp-media.c:
243 rtsp-media: Only count senders when counting blocked streams
244 Only sender streams sends the GstRTSPStreamBlocking message, so only
245 these should be counted before setting media status to prepared.
246 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
248 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
250 * gst/rtsp-sink/gstrtspclientsink.c:
251 rtspclientsink add proper support for uri queries
252 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
254 2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
256 * gst/rtsp-server/rtsp-client.c:
257 rtsp-client: Only unref client watch context on finalize, to avoid deadlock
258 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
259 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
261 2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
263 * gst/rtsp-server/rtsp-stream.c:
264 rtsp-stream: collect a clock_rate when blocking
265 This lets us provide a clock_rate in a fashion similar to the
266 other code paths in get_rtpinfo()
267 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
269 2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
271 * gst/rtsp-server/rtsp-media.c:
272 rtsp-media: Use guint64 for setting the size-time property on rtpstorage
273 Otherwise this will cause memory corruption as the property expects a 64
275 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
277 2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
279 * gst/rtsp-server/rtsp-media.c:
280 * gst/rtsp-server/rtsp-stream.c:
281 rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
282 To prevent cases with prerolling when the inactive stream prerolls first
283 and the server proceeds without waiting for the active stream, we will
284 ignore GstRTSPStreamBlocking messages from incomplete streams. When
285 there are no complete streams (during DESCRIBE), we will listen to all
287 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
289 2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
291 * tests/check/gst/media.c:
292 * tests/check/meson.build:
293 * tests/files/test.avi:
294 media test: Add test for seeking one active stream with a demuxer
295 Add another seek_one_active_stream test but with a demuxer. The demuxer
296 will flush both streams in opposed to the existing test which only
297 flushes the active stream. This will help exposing problems with the
298 prerolling process after a flushing seek.
299 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
301 2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
303 * gst/rtsp-server/meson.build:
305 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
306 * pkgconfig/gstreamer-rtsp-server.pc.in:
307 * pkgconfig/meson.build:
308 Meson: Use pkg-config generator
309 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
311 2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
314 meson: update glib minimum version to 2.56
315 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
317 2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
319 * examples/test-launch.c:
320 * gst/rtsp-server/rtsp-media-factory.c:
321 * gst/rtsp-server/rtsp-media-factory.h:
322 * gst/rtsp-server/rtsp-media.c:
323 * gst/rtsp-server/rtsp-server-internal.h:
324 * gst/rtsp-server/rtsp-stream.c:
325 * tests/check/gst/client.c:
326 rtsp-media-factory: expose API to disable RTCP
327 This is supported by the RFC, and can be useful on systems where
328 allocating two consecutive ports is problematic, and RTCP is not
330 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
332 2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
334 * hooks/pre-commit.hook:
336 git: use our standard pre commit hook
337 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
339 2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
341 * gst/rtsp-server/rtsp-stream.c:
342 rtsp-stream: make use of blocked_running_time in query_position
343 When blocking, the sink element will not have received a buffer
344 yet and the position query will fail. Instead, we make use of
345 the running time of the buffer we blocked on.
346 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
348 2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
350 * gst/rtsp-server/rtsp-stream.c:
351 rtsp-stream: collect rtp info when blocking
352 We don't unblock the stream anymore before replying to the
353 play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
354 so the sinks don't have a last-sample after potentially flush
355 seeking. seek_trickmode waits for preroll however, which means
356 the stream will block and wait for a first buffer. Subsequent
357 calls to get_rtpinfo() can thus make use of the information.
358 See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
359 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
361 2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
363 * examples/meson.build:
364 * examples/test-replay-server.c:
365 * examples/test-replay-server.h:
366 examples: Add an example for loop playback
367 This demo example shows a way of file loop playback of a given source.
368 Note that client seek request is not properly implemented yet.
369 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
371 2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
373 * gst/rtsp-server/rtsp-media.c:
374 rtsp-media: Plug memory leak
375 The get-storage signal of rtpbin increases the ref count of the storage.
376 So we have to unref it after usage.
377 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
379 2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
381 * gst/rtsp-server/rtsp-media.c:
382 rtsp-media: Get rates only on sender streams
383 When play a media with both sender and receiver stream, like ONVIF
384 back channel audio in, gst_rtsp_media_get_rates call
385 gst_rtsp_stream_get_rates for each stream to set the rates. But
386 gst_rtsp_stream_get_rates return false for the receiver steam, which
387 lead a g_assert crash.
388 Instead to get rates on all streams, now just get rates on sender
390 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
392 2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
394 * gst/rtsp-server/rtsp-media.c:
395 * gst/rtsp-server/rtsp-server-internal.h:
396 * gst/rtsp-server/rtsp-stream.c:
397 rtsp-media: set a 0 storage size for TCP receivers
398 ulpfec correction is obviously useless when receiving a stream
399 over TCP, and in TCP modes the rtp storage receives non
400 timestamped buffers, causing it to queue buffers indefinitely,
401 until the queue grows so large that sanity checks kick in and
402 warnings start to get emitted.
403 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
405 2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
407 * gst/rtsp-server/rtsp-stream.c:
408 rtsp-stream: preroll on gap events
409 This allows negotiating a SDP with all streams present, but only
410 start sending packets at some later point in time
411 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
413 2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
415 * gst/rtsp-server/rtsp-media.c:
416 rtsp-media: do not unblock on unsuspend
417 rtsp_media_unsuspend() is called from handle_play_request()
418 before sending the play response. Unblocking the streams here
419 was causing data to be sent out before the client was ready
420 to handle it, with obvious side effects such as initial packets
421 getting discarded, causing decoding errors.
422 Instead we can simply let the media streams be unblocked when
423 the state of the media is set to PLAYING, which occurs after
424 sending the play response.
425 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
427 2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
430 ci: include template from gst-ci master branch again
432 2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
434 * docs/gst_plugins_cache.json:
438 === release 1.18.0 ===
440 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
446 * docs/gst_plugins_cache.json:
447 * gst-rtsp-server.doap:
451 === release 1.17.90 ===
453 2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
458 * docs/gst_plugins_cache.json:
459 * gst-rtsp-server.doap:
463 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
465 * gst/rtsp-server/rtsp-thread-pool.c:
466 rtsp-thread-pool.c: fix clang 10 warning
467 clang 10 is complaining about incompatible types due to the
470 ../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]
472 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
474 2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
476 * gst/rtsp-server/rtsp-thread-pool.c:
477 rtsp-thread-pool.c: fix clang 10 warning
478 clang 10 is complaining about incompatible types due to the
481 ../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]
483 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
485 2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
487 * gst/rtsp-server/rtsp-sdp.c:
488 rtsp-sdp: Fix resource leak in mikey messsage
489 Fixed a resource leak for mikey message while adding crypto session
491 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
493 2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
496 * scripts/extract-release-date-from-doap-file.py:
497 meson: set release date from .doap file for releases
498 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
500 2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
502 * gst/rtsp-server/rtsp-stream.c:
503 rtsp-stream: explicitly set caps on udpsrc elements
504 This causes them to send caps events before data flow, which is
505 usually a pretty correct thing to do!
506 Not doing so manifested in a bug where ssrcdemux wouldn't forward
507 the caps it had received with an extra ssrc field, as it hadn't
508 received any caps event.
510 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
512 2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
514 * docs/gst_plugins_cache.json:
518 === release 1.17.2 ===
520 2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
525 * docs/gst_plugins_cache.json:
526 * gst-rtsp-server.doap:
530 2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
532 * docs/gst_plugins_cache.json:
533 doc: Stop documenting properties from parents
535 2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
537 * docs/gst_plugins_cache.json:
538 docs: Fix version in the plugins cache
539 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
541 2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
543 * gst/rtsp-sink/gstrtspclientsink.c:
544 rtspclientsink: Don't call gst_ghost_pad_construct() anymore
545 It's deprecated, unneeded and doesn't do anything anymore.
546 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
548 2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
553 === release 1.17.1 ===
555 2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
560 * docs/gst_plugins_cache.json:
561 * gst-rtsp-server.doap:
565 2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
567 * gst/rtsp-server/rtsp-media.c:
568 rtsp-media: Add/configure transports when completing the pipeline
569 Otherwise the transports are not set up yet during the PLAY request
570 handling when unsuspending (and thus unblocking) the media.
571 In case of live pipelines this then causes the first few packets to go
572 to the sinks before they know what to do with them, and they simply
573 discard them which is rather suboptimal in case of keyframes.
574 For non-live pipelines this is not a problem because the sink will still
575 be PAUSED and as such not send out the data yet but wait until it goes
576 to PLAYING, which is late enough.
577 Adding the transports multiple times is not a problem: if the transport
578 is already added it won't be added another time and TRUE will be
580 This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
582 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
583 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
585 2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
587 * gst/rtsp-server/rtsp-media.c:
588 rtsp-media: Fix misleading comment
589 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
591 2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
593 * gst/rtsp-server/rtsp-media.c:
594 rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
595 The pad probes are not needed anymore at this point and later when
596 reaching buffering 100% only the state is changed, no unblocking
598 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
600 2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
602 * gst/rtsp-server/rtsp-media.c:
603 rtsp-media: Remove duplicated media_unblock() function
604 It does literally the same as media_streams_set_blocked(FALSE).
605 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
607 2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
609 * examples/test-onvif-server.c:
610 test-onvif-server: cast ntp-offset property value to 64 bit
611 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
613 2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
615 * docs/gst_plugins_cache.json:
616 docs: Update plugins cache
618 2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
620 * examples/test-onvif-server.c:
621 * examples/test-onvif-server.h:
622 * gst/rtsp-server/rtsp-onvif-media-factory.h:
623 onvif-media-factory: define autoptr cleanup function
624 And have the factory in the onvif-server example inherit from
625 GstRTSPOnvifMediaFactory.
626 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
628 2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
630 * docs/gst_plugins_cache.json:
631 docs: Update plugins cache
633 2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
635 * tests/check/gst/rtspserver.c:
636 tests: enforce I420 format
637 Test was not enforcing a video format on videotestsrc. I420 was picked as it
638 was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
639 true (gst-plugins-base!689).
640 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
642 2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
644 * gst/rtsp-sink/gstrtspclientsink.c:
645 plugins: uddate gst_type_mark_as_plugin_api() calls
647 2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
650 doc: Require hotdoc >= 0.11.0
652 2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
654 * docs/gst_plugins_cache.json:
655 docs: Update gst_plugins_cache.json
657 2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
659 * gst/rtsp-sink/gstrtspclientsink.c:
660 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
662 2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
664 * gst/rtsp-server/meson.build:
665 meson: gir: remove bogus sources_top_dir kwarg
666 Doesn't actually exist. Was fixed differently in Meson
667 so that the user doesn't have to specify it.
668 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
670 2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
672 * tests/check/meson.build:
673 tests: put registry into tests/check not the gst/ subdir
674 Underscorify the test name before setting GST_REGISTRY,
675 so the registry actually ends up in the current build dir
677 For consistency with the other modules, but should also
678 avoid problems on windows.
679 Also fix indentation of environment block.
680 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
682 2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
684 * tests/check/meson.build:
685 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
686 If core is built as a subproject (e.g. as in gst-build), make sure to use
687 the gst-plugin-scanner from the built subproject. Without this, gstreamer
688 might accidentally use the gst-plugin-scanner from the install prefix if
689 that exists, which in turn might drag in gst library versions we didn't
690 mean to drag in. Those gst library versions might then be older than
691 what our current build needs, and might cause our newly-built plugins
692 to get blacklisted in the test registry because they rely on a symbol
693 that the wrongly-pulled in gst lib doesn't have.
694 This should fix running of unit tests in gst-build when invoking
695 meson test or ninja test from outside the devenv for the case where
696 there is an older or different-version gst-plugin-scanner installed
697 in the install prefix.
698 In case no gst-plugin-scanner is installed in the install prefix, this
699 will fix "GStreamer-WARNING: External plugin loader failed. This most
700 likely means that the plugin loader helper binary was not found or
701 could not be run. You might need to set the GST_PLUGIN_SCANNER
702 environment variable if your setup is unusual." warnings when running
704 In the case where we find GStreamer core via pkg-config we use
705 a newly-added pkg-config var "pluginscannerdir" to get the right
706 directory. This has the benefit of working transparently for both
707 installed and uninstalled pkg-config files/setups.
708 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
710 2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
712 * tests/check/meson.build:
713 tests: gst-plugins-base and -bad plugins are required for the unit tests
714 Make hard requirement until we have more fine-grained control
715 in the unit tests. Of course the presence of the .pc file doesn't
716 imply that the plugins we need are actually there, but it's at
717 least a step in the right direction.
718 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
720 2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
722 * tests/check/meson.build:
723 tests: pick up rtsp-server plugins from build directory only
724 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
726 2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
728 * gst/rtsp-server/rtsp-media.c:
729 rtsp-media: wait for all GstRTSPStreamBlocking messages
730 Make sure rtsp-media have received a GstRTSPStreamBlocking message from
731 each active stream when checking if all streams are blocked.
732 Without this change there will be a race condition when using two or
733 more streams and rtsp-media receives a GstRTSPStreamBlocking message
734 from one of the streams. This is because rtsp-media then checks if all
735 streams are blocked by calling gst_rtsp_stream_is_blocking() for each
736 stream. This function call returns TRUE if the stream has sent a
737 GstRTSPStreamBlocking message, however, rtsp-media may have yet to
738 receive this message. This would then result in that rtsp-media
739 erroneously thinks it is blocking all streams which could result in
740 rtsp-media changing state, from PREPARING to PREPARED. In the case of a
741 preroll, this could result in that rtsp-media thinks that the pipeline
742 is prerolled even though that might not be the case.
743 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
745 2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
747 * gst/rtsp-server/rtsp-media.c:
748 rtsp-media: update expected_async_done during suspend
749 Set expected_async_done to FALSE in default_suspend() if a state change
750 occurs and the return value from set_target_state() is something other
751 than GST_STATE_CHANGE_ASYNC.
752 Without this change there is a risk that expected_async_done will be
753 TRUE even though no asynchronous state change is taking place. This
754 could happen if the pipeline is set to PAUSED using
755 media_set_pipeline_state_locked(), an asynchronous state change starts
756 and then the media is suspended (which could result in a state change,
757 aborting the asynchronous state change). If the media is suspended
758 before the asynchronous state change ends then expected_async_done will
759 be TRUE but no asynchronous state change is taking place.
760 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
762 2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
764 * gst/rtsp-server/rtsp-client.c:
765 rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
766 There was a race condition where client was being finalized and
767 concurrently in some other thread the rtsp ctrl timout was relying on
768 client data that was being freed.
769 When rtsp ctrl timeout is setup, a WeakRef on Client is set.
770 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
772 2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
774 * gst/rtsp-server/rtsp-media-factory.c:
775 * gst/rtsp-server/rtsp-media-factory.h:
776 * gst/rtsp-server/rtsp-media.c:
777 * gst/rtsp-server/rtsp-media.h:
778 media-factory: complete DSCP QoS setting support
779 add dscp_qos setting support at factory and media level to setup IP DSCP
780 field of bounded UDP sinks.
781 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
782 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
784 2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
786 * gst/rtsp-server/rtsp-client.c:
787 rtsp-client: Fix some race conditions around timeout source removal
788 We always need to take the lock while accessing it as otherwise another
789 thread might've removed it in the meantime. Also when destroying and
790 creating a new one, ensure that the mutex is not shortly unlocked in
791 between as during that time another one might potentially be created
793 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
795 2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
797 * gst/rtsp-server/rtsp-media.c:
798 * gst/rtsp-server/rtsp-stream.c:
799 rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
800 And the same for gst_rtsp_stream_get_rates().
801 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
803 2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
805 * examples/test-onvif-server.c:
806 examples: test-onvif-server: fix compiler warnings on raspbian
807 Fix printf format for 64-bit variables.
808 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
810 2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
812 * gst/rtsp-server/rtsp-stream-transport.c:
813 * gst/rtsp-server/rtsp-stream-transport.h:
814 * gst/rtsp-server/rtsp-stream.c:
815 rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
816 The old API is preserved now and new API was added that provides the
817 additional parameter to the callback.
818 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
819 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
821 2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
823 * gst/rtsp-server/rtsp-client.c:
824 rtsp-client: Store the timeout source by pointer instead of id
825 That way we don't have to retrieve it again from the main context when
826 destroying it but can directly do so.
827 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
829 2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
831 * gst/rtsp-server/rtsp-client.c:
832 rtsp-client: Clean up watch/watch context and related state consistently
833 And assert that it was cleaned up properly before the client is
834 finalized. If something is still around when the client is shut down
835 then something went very wrong before.
836 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
838 2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
840 * gst/rtsp-server/rtsp-client.c:
841 * tests/check/gst/rtspserver.c:
842 rtsp-client: Combine the pre-session and post-session timeout
843 They previously used the same state but different mechanisms and
844 functions, which was difficult to follow, error prone and simply
846 Also adjust the test for the post-session timeout a bit to be less racy
847 now that the timing has slightly changed.
848 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
850 2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
852 * gst/rtsp-server/rtsp-client.c:
853 rtsp-client: Don't ever close the client connection directly when a session is torn down
854 There might be other sessions that are running over the same RTSP
855 connection and we should not simply close the client directly if one of
857 By default the connection will be closed once the client closes it or
858 the OS does. This behaviour can be adjusted with the
859 post-session-timeout property, which allows to close it automatically
860 from the server side after all sessions are gone and the given timeout
862 This reverts the previous commit.
863 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
865 2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
867 * gst/rtsp-server/rtsp-client.c:
868 rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
869 Instead of closing it never at all. Previously there was only code that
870 closed the client asynchronously if sending the response happened
871 asynchrously at a later time.
872 Thanks to Christian M for debugging this issue.
873 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
874 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
876 2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
878 * gst/rtsp-server/rtsp-stream.c:
879 rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
880 Otherwise no sink is found for multicast sreams and the less accurate
881 fallback is used to determine the current sequence number and timestamp.
883 2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
885 * gst/rtsp-server/rtsp-auth.c:
886 rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
887 When using the basic authentication scheme, we wouldn't validate that
888 the authorization field of the credentials is not NULL and pass it on
889 to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
890 dereference the NULL pointer and crash.
891 A specially crafted (read: invalid) RTSP header can cause this to
893 As a solution, check for the authorization to be not NULL before
894 continuing processing it and if it is simply fail authentication.
895 This fixes CVE-2020-6095 and TALOS-2020-1018.
896 Discovered by Peter Wang of Cisco ASIG.
898 2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
900 * gst/rtsp-server/rtsp-client.c:
901 rtsp-client: Use watch_context before unref
902 Move the usage of priv->watch_context to beginning of function
903 gst_rtsp_client_finalize. Instead of use it after
904 g_main_context_unref (priv->watch_context).
906 2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
908 * gst/rtsp-server/rtsp-stream.c:
909 rtsp-stream: fix deadlock on transport removal
910 We cannot take the RTSPStream lock while holding a transport backlog
911 lock, as remove_transport may be called externally, which will
912 take first the RTSPStream lock then the transport backlog lock.
914 2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
916 * gst/rtsp-server/rtsp-server-internal.h:
917 * gst/rtsp-server/rtsp-stream-transport.c:
918 * gst/rtsp-server/rtsp-stream.c:
919 rtsp-stream: clear backlog when removing transport
920 This ensures we don't end up calling any of transports' callbacks
921 with a potentially unreffed user_data (in practice, a client that
922 may have been removed)
924 2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
926 * gst/rtsp-server/rtsp-stream.c:
927 rtsp-stream: marshal calls to send_tcp_message to a single thread
928 In order to address the race condition pointed out at
929 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
930 we get rid of the send thread pool, and instead spawn and manage
931 a single thread to pull samples from app sinks and add them to
932 the transport's backlogs.
933 Additionally, we now also always go through the backlogs in order
934 to simplify the logic.
936 2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
938 * gst/rtsp-server/rtsp-server-internal.h:
939 * gst/rtsp-server/rtsp-stream-transport.c:
940 * gst/rtsp-server/rtsp-stream.c:
941 rtsp-stream: properly protect TCP backlog access
943 We cannot hold stream->lock while pushing data, but need
944 to consistently check the state of the backlog both from
945 the send_tcp_message function and the on_message_sent function,
946 which may or may not be called from the same thread.
947 This commit introduces internal API to allow for potentially
948 recursive locking of transport streams, addressing a race
949 condition where the RTSP stream could push items out of order
950 when popping them from the backlog.
952 2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
954 * gst/rtsp-server/rtsp-media.c:
955 rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
956 It's taken ownership of by the media, and returned with `transfer none`
957 from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
958 first then any bindings will wrongly take ownership of the pipeline once
959 it arrives in bindings code.
961 2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
963 * examples/test-onvif-client.c:
964 Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
966 2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
968 * gst/rtsp-server/rtsp-media.c:
969 rtsp-media: fix default latency
971 2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
973 * gst/rtsp-server/rtsp-client.c:
974 rtsp-client: make closing more thread safe
975 + Take the watch lock prior to using priv->watch
976 + Flush both the watch and connection before closing / unreffing
977 gst_rtsp_connection_close() is not threadsafe on its own, this is
978 a workaround at the client level, where we control both the watch
981 2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
983 * gst/rtsp-server/rtsp-latency-bin.c:
984 rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
987 Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
988 `your_type_get_instance_private()` function instead
991 2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
993 * gst/rtsp-server/rtsp-client.c:
994 * tests/check/gst/rtspserver.c:
995 rtsp-client: add property post-session-timeout
996 This is a TCP connection timeout for client connections, in seconds.
997 If a positive value is set for this property, the client connection
998 will be kept alive for this amount of seconds after the last session
999 timeout. For negative values of this property the connection timeout
1000 handling is delegated to the system (just as it was before).
1003 2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
1005 * gst/rtsp-server/rtsp-stream.c:
1006 rtsp-stream: check for NULL transports prior to ref'ing
1008 2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1010 * gst/rtsp-server/rtsp-server-internal.h:
1011 * gst/rtsp-server/rtsp-stream-transport.c:
1012 * gst/rtsp-server/rtsp-stream.c:
1013 rtsp-stream: fix checking of TCP backpressure
1014 The internal index of our appsinks, while it can be used to
1015 determine whether a message is RTP or RTCP, is not necessarily
1016 the same as the interleaved channel. Let the stream-transport
1017 determine the channel to check backpressure for, the same way
1018 it determines the channel according to whether it is sending
1021 2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
1023 * gst/rtsp-server/rtsp-session.c:
1024 rtsp-session: Butcher the file to please gst-indent in the CI
1025 This should be reverted once the CI has an updated gst-indent.
1027 2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
1029 * gst/rtsp-server/rtsp-session.c:
1030 * gst/rtsp-server/rtsp-session.h:
1031 * gst/rtsp-sink/gstrtspclientsink.c:
1032 * gst/rtsp-sink/gstrtspclientsink.h:
1033 rtsp-session & client: Remove deprecated GTimeVal
1034 GTimeVal won't work past 2038
1036 2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
1038 * gst/rtsp-server/rtsp-auth.c:
1039 rtsp-auth: fix default token leak
1041 2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
1043 * gst/rtsp-sink/gstrtspclientsink.c:
1044 gstrtspclientsink: unref transports when closing bin
1047 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1049 * gst/rtsp-server/rtsp-media.c:
1050 rtsp-media: Force seek when flush flag is set
1051 The commit "rtsp-client: define all seek accuracy flags from
1052 setup_play_mode" changed the behaviour of when doing a seek.
1053 Before that commit, having the flush flag set would result in a seek
1055 Even if no seek was needed. One reason to force seek is to flush old buffers
1056 created in Describe requests.
1057 Thus adding force seek also for flush flag will result in play request
1060 2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
1062 * gst/rtsp-server/rtsp-client.c:
1063 rtsp-client: Revitalize dead code
1064 Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
1067 2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
1069 * gst/rtsp-server/rtsp-sdp.c:
1070 rtsp-sdp: Don't try to use non-initialized values
1071 Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
1072 returns TRUE. Also avoid the whole clock signalling block if we're not
1073 dealing with senders.
1078 2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
1080 * gst/rtsp-server/rtsp-stream-transport.c:
1081 * gst/rtsp-server/rtsp-stream.c:
1082 * tests/check/gst/stream.c:
1083 rtsp-stream: Removing invalid transports returns false
1084 When removing transports an assertion was that the transports passed in
1085 for removal are present in the list, however that can't be assumed.
1086 As an example if a transport was removed from a thread running
1087 send_tcp_message, the main thread can try to remove the same transport
1088 again if it gets a handle_pause_request. This will not effect the
1089 transport list but it will effect n_tcp_transports as it will be
1090 decrement and then have the wrong value.
1092 2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
1094 * tests/check/gst/client.c:
1095 client test: add scale and speed negative tests
1096 Negative tests for scale and speed should be done as well, verify that
1097 the response code is "400 Bad request" when a bad request is done.
1099 2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
1101 * gst/rtsp-server/rtsp-auth.c:
1102 * gst/rtsp-server/rtsp-client.c:
1103 * gst/rtsp-server/rtsp-media-factory.c:
1104 * gst/rtsp-server/rtsp-media.c:
1105 * gst/rtsp-server/rtsp-server.c:
1106 * gst/rtsp-server/rtsp-session-pool.c:
1107 * gst/rtsp-server/rtsp-stream.c:
1108 * gst/rtsp-sink/gstrtspclientsink.c:
1109 Don't pass default GLib marshallers for signals
1110 By passing NULL to `g_signal_new` instead of a marshaller, GLib will
1111 actually internally optimize the signal (if the marshaller is available
1112 in GLib itself) by also setting the valist marshaller. This makes the
1113 signal emission a bit more performant than the regular marshalling,
1114 which still needs to box into `GValue` and call libffi in case of a
1116 Note that for custom marshallers, one would use
1117 `g_signal_set_va_marshaller()` with the valist marshaller instead.
1119 2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1121 * gst/rtsp-server/rtsp-mount-points.c:
1122 GstRTSPMountPoints: Remove any existing factory before adding a new one
1123 The documentation of gst_rtsp_mount_points_add_factory() says "Any
1124 previous mount point will be freed" which was true when it was
1125 implemented using a GHashTable. But in 2012 it got rewrote using a
1126 GSequence and since then it could have 2 factories for the same path.
1127 Which one gets used is random, depending on the sorting order of 2
1130 2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1132 * gst/rtsp-server/rtsp-client.c:
1133 * gst/rtsp-server/rtsp-server-internal.h:
1134 * gst/rtsp-server/rtsp-stream-transport.c:
1135 * gst/rtsp-server/rtsp-stream-transport.h:
1136 * gst/rtsp-server/rtsp-stream.c:
1137 stream: refactor TCP backpressure handling
1138 The previous implementation stopped sending TCP messages to
1139 all clients when a single one stopped consuming them, which
1140 obviously created problems for shared media.
1141 Instead, we now manage a backlog in stream-transport, and slow
1142 clients are removed once this backlog exceeds a maximum duration,
1143 currently hardcoded.
1146 2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1149 meson: build gir even when cross-compiling if introspection was enabled explicitly
1150 This can be made to work in certain circumstances when
1151 cross-compiling, so default to not building g-i stuff
1152 when cross-compiling, but allow it if introspection was
1153 enabled explicitly via -Dintrospection=enabled.
1154 See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
1156 2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
1158 * gst/rtsp-server/rtsp-session.c:
1159 rtsp-session: clean up comment extra-timeout
1161 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
1163 * gst/rtsp-server/rtsp-client.c:
1164 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
1165 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
1166 from the RTSP context.
1169 2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1171 * gst/rtsp-server/rtsp-client.c:
1172 * gst/rtsp-server/rtsp-media.c:
1173 * gst/rtsp-server/rtsp-media.h:
1174 rtsp-client: Lock shared media
1175 For shared media we got race conditions. Concurrently rtsp clients might
1176 suspend or unsuspend the shared media and thus change the state without
1177 the clients expecting that.
1178 By introducing a lock that can be taken by callers such as rtsp_client
1179 one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
1180 to handle the media sequentially thus allowing one client to finish its
1181 rtsp call before another client calls on the same media.
1182 https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
1185 2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
1187 * gst/rtsp-server/rtsp-session.c:
1188 rtsp-session: add property extra-timeout
1189 Extra time to add to the timeout, in seconds. This only
1190 affects the time until a session is considered timed out
1191 and is not signalled in the RTSP request responses.
1192 Only the value of the timeout property is signalled in the
1195 2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
1197 * gst/rtsp-server/rtsp-stream.c:
1198 rtsp-stream : fix race condition in send_tcp_message
1199 If one thread is inside the send_tcp_message function and are done
1200 sending rtp or rtcp messages so the n_outstanding variable is zero
1201 however have not exit the loop sending the messages. While sending its
1202 messages, transports have been added or removed to the transport list,
1203 so the cache should be updated. If now an additional thread comes to
1204 the function send_tcp_message and trying to send rtp messages it will
1205 first destroy the rtp cache that is still being iterated trough by the
1209 2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
1218 * examples/.gitignore:
1219 * examples/Makefile.am:
1221 * gst/rtsp-server/.gitignore:
1222 * gst/rtsp-server/Makefile.am:
1223 * gst/rtsp-sink/Makefile.am:
1224 * pkgconfig/.gitignore:
1225 * pkgconfig/Makefile.am:
1227 * tests/Makefile.am:
1228 * tests/check/Makefile.am:
1229 Remove autotools build
1231 Maybe we can now use the meson pkgconfig module
1232 for .pc files? (Does it support uninstalled now?)
1234 2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
1236 * tests/check/gst/client.c:
1237 client: fix test mem leak in attach_rate_tweaking_probe
1239 2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
1241 * tests/check/gst/media.c:
1242 media: remove memleak in test test_media_seek
1244 2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
1246 * tests/check/gst/rtspserver.c:
1247 rtspserver: Remove memleak in test test_double_play
1249 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
1251 * gst/rtsp-server/rtsp-media.c:
1252 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
1254 2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
1256 * gst/rtsp-server/rtsp-media.c:
1257 * tests/check/gst/rtspserver.c:
1258 rtsp-media: Unblock all streams
1259 When unsuspending and going to PLAYING, unblock all streams instead of
1260 only those that are linked (the linked streams are the ones for which
1261 SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
1262 pushing buffers on unlinked streams.
1263 This change is because playback using single-threaded demuxers like
1264 matroska-demux could be blocked if SETUP was not called for all media.
1265 Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
1266 gstflvdemux, qtdemux, and matroska-demux) will handle
1267 GST_FLOW_NOT_LINKED automatically.
1270 2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
1272 * gst/rtsp-server/rtsp-media.c:
1273 * tests/check/gst/rtspserver.c:
1274 rtsp-media: Wait on async when needed.
1275 Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
1276 In the unit test the pause from adjust_play_mode will cause a preroll
1277 and after that async-done will be produced.
1278 Without this patch there are no one consuming this async-done and when
1279 later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
1280 wait for async-done. But then it wrongly find the async-done prodused by
1281 adjus_play_mode and continue executing without waiting for the preroll
1284 2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
1286 * gst/rtsp-server/rtsp-client.c:
1287 rtsp-client: RTP Info when completed_sender
1288 Change condition that should be fulfilled regarding RTPInfo.
1289 Replace !gst_rtsp_media_is_receive_only with
1290 gst_rtsp_media_has_completed_sender. It is more correct to actually look
1291 for a sender pipeline that is complete. Only then a RTPInfo should
1293 gst_rtsp_media_is_receive_only gives different answears depending on
1295 If Describe is called wth URL+options for backchannel SDP will give only
1296 audio and only backchannel a=sendonly
1297 If Describe is called on URL+options that gives both audio and video
1298 direction from server to client, pipelines are created. Thus
1299 receive_only will return false, even though Setup only would setup
1301 RTP-Info is only for outgoing streams. Thus one should look if outgoing
1302 streams are complete.
1304 2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
1306 * gst/rtsp-server/rtsp-client.c:
1307 * tests/check/gst/client.c:
1308 rtsp-client: RTP Info exists conditionally in PLAY
1309 If RTP Info is missing and it is not a receiver only, eg. audio
1310 backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
1311 In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
1312 Since 1.14 there is audio backchannel support. Thus RTP-info is
1313 conditional now. When audio backchannel only mode, there is no RTP-info.
1316 2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1318 * examples/test-onvif-client.c:
1319 test-onvif-client: remove unused query
1321 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
1323 * gst/rtsp-server/rtsp-client.c:
1324 rtsp-client: RTP Info must exist in PLAY response
1325 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
1328 2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1330 * examples/test-onvif-client.c:
1331 test-onvif-client: perform accurate seeks
1332 See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
1333 Also, modify how we compute the position: position queries in
1334 PAUSED mode fail to account for the newly-prerolled frame, leading
1335 to frame skips when performing seeks in that state. Instead,
1336 compute the current position from the last sample.
1338 2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
1340 * gst/rtsp-server/rtsp-client.c:
1341 * gst/rtsp-server/rtsp-media.c:
1342 * gst/rtsp-server/rtsp-media.h:
1343 * tests/check/gst/rtspserver.c:
1344 Use complete streams for scale and speed.
1345 Without this patch it's always stream0 that is used to get segment event
1346 that is used to set scale and speed. This even if client not doing SETUP
1347 for stream0. At least in suspend mode reset this not working since then
1348 it's just random if send_rtp_sink have got any segment event. There are
1349 no check if send_rtp_sink for stream0 got any data before media is
1350 prerolled after PLAY request.
1352 2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
1354 * examples/test-onvif-server.c:
1355 * examples/test-onvif-server.h:
1356 examples/onvif-server: fix werror build with clang
1357 ../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]
1358 self->incoming_segment->format, self->incoming_segment->flags,
1359 ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
1360 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
1361 G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
1363 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1364 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1366 <scratch space>:77:1: note: expanded from here
1369 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
1370 G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
1372 /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1373 static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
1375 <scratch space>:9:1: note: expanded from here
1378 ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
1379 /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
1380 static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
1382 <scratch space>:12:1: note: expanded from here
1386 2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
1389 meson: Don't generate doc cache when no plugins are enabled
1390 Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
1392 2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1394 * examples/test-onvif-client.c:
1395 test-onvif-client: stdin is not defined in MSVC
1397 2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1399 * gst/rtsp-server/rtsp-media.c:
1400 rtsp-media: add missing Since tag
1402 2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1404 * examples/test-onvif-client.c:
1405 test-onvif-client: STDIN_FILENO is not portable
1406 If not defined, define it to _fileno(stdin) on Windows, 0
1409 2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1411 * examples/test-onvif-server.c:
1412 test-onvif-server: downgrade logging
1414 2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1416 * examples/meson.build:
1417 * examples/test-onvif-client.c:
1418 * examples/test-onvif-server.c:
1419 examples: add ONVIF client / server example
1421 2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1423 * gst/rtsp-server/rtsp-client.c:
1424 * gst/rtsp-server/rtsp-media.c:
1425 rtsp-client: define all seek accuracy flags from setup_play_mode
1426 We then pass those to adjust_play_mode, which needs to operate
1427 on the "final" seek flags, as previously the code in rtsp-media
1428 was assuming that accuracy seek flags (accurate / key_unit) should
1429 not be set if the flags passed to the seek method were already set.
1431 2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
1433 * gst/rtsp-server/rtsp-media-factory-uri.c:
1434 * gst/rtsp-server/rtsp-media.c:
1435 rtsp-media: Try to get dynamic payloaders by name from their bin first
1436 First try "pay", then "pay_%s" (where %s == pad name). And only then
1437 fall back to the code that simply takes the first payloader that is
1439 The current code usually works (but is racy) because it will always take
1440 the payloader that was last added (due to g_list_prepend() when adding
1441 elements) in pad-added and that's usually the correct one. But if a new
1442 payloader is added between pad-added and us trying to get it, we would
1443 get the wrong payloader.
1445 2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1447 * tests/check/gst/client.c:
1448 client test: expect any port in transport
1449 setup_multicast_client sets a 5000-5010 range for the client
1450 ports, it is incorrect to expect the transport to always use
1454 2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1456 * tests/check/gst/onvif.c:
1457 onvif tests: use g_cond_wait() correctly
1458 g_cond_wait() has to be called in a loop until required conditions
1462 2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
1464 * gst/rtsp-server/rtsp-stream.c:
1465 rtsp-stream: Not wait on receiver streams when pre-rolling
1466 Without this patch there are problem pre-rolling when using audio back
1468 Without this patch a probe will be created for all streams including
1469 the stream for audio backchannel. To pre-roll all this pads have to
1470 receive data. Since the stream for audio backchannel is a receiver this
1472 The solution is to never create any probes for streams that are for
1473 incomming data and instead set them as blocking already from beginning.
1475 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
1477 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1478 * gst/rtsp-server/rtsp-onvif-media.c:
1479 onvif-media: fix "void function returning a value" compiler warning
1481 2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1483 * gst/rtsp-server/rtsp-media.c:
1484 rtsp-media: make sure streams are blocked when sending seek
1485 The recent ONVIF work exposed a race condition when dealing with
1486 multiple streams: one of the sinks may preroll before other streams
1487 have started flushing. This led to the pipeline posting async-done
1488 prematurely, when some streams were actually still in the middle
1489 of performing a flushing seek. The newly-added code looks up a
1490 sticky segment event on the first stream in order to respond to
1491 the PLAY request with accurate Scale and Speed headers. In the
1492 failure condition, the first stream was flushing, and thus had
1493 no sticky segment event, leading to the PLAY request failing,
1494 and in turn the test.
1496 2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
1499 * gst/rtsp-server/rtsp-media-factory-uri.h:
1502 2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1504 * gst/rtsp-server/rtsp-client.c:
1505 * gst/rtsp-server/rtsp-client.h:
1506 * gst/rtsp-server/rtsp-media.c:
1507 * gst/rtsp-server/rtsp-media.h:
1508 * gst/rtsp-server/rtsp-onvif-client.c:
1509 * gst/rtsp-server/rtsp-onvif-client.h:
1510 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1511 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1512 * gst/rtsp-server/rtsp-onvif-media.c:
1513 * gst/rtsp-server/rtsp-onvif-server.h:
1514 * gst/rtsp-server/rtsp-stream.c:
1515 * gst/rtsp-server/rtsp-stream.h:
1516 * tests/check/gst/media.c:
1517 * tests/check/gst/onvif.c:
1518 * tests/check/meson.build:
1519 onvif: Implement and test the Streaming Specification
1520 https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
1522 2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1524 * gst/rtsp-server/rtsp-client.c:
1525 * gst/rtsp-server/rtsp-client.h:
1526 rtsp-client: add gst_rtsp_client_get_stream_transport()
1527 This will be used in the onvif tests in order to validate the
1528 data transmitted over TCP: for streaming to continue after a
1529 data message has been provided to client->send_func, the client
1530 is responsible for marking the message as sent on the relevant
1533 2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1535 * gst/rtsp-server/rtsp-client.c:
1536 client: Scale implies TRICK_MODE
1538 2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1540 * gst/rtsp-server/rtsp-client.c:
1541 client: compare booleans, not pointers to them
1543 2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
1545 * gst/rtsp-server/rtsp-media.c:
1546 * gst/rtsp-server/rtsp-stream.c:
1547 * tests/check/gst/media.c:
1548 Reverse playback support
1549 GStreamer plays segment from stop to start when doing reverse playback.
1550 RTSP implies that media should be played from start of Range header to
1551 its stop. Hence we swap start and stop times before passing them to
1553 Also make gst_rtsp_stream_query_stop always return value that can be
1554 used as stop time of Range header.
1556 2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
1558 * gst/rtsp-server/rtsp-client.c:
1559 * gst/rtsp-server/rtsp-media.c:
1560 * gst/rtsp-server/rtsp-media.h:
1561 * tests/check/gst/client.c:
1562 rtsp-client: add support for Scale and Speed header
1563 Add support for the RTSP Scale and Speed headers by setting the rate in
1564 the seek to (scale*speed). We then check the resulting segment for rate
1565 and applied rate, and use them as values for the Speed and Scale headers
1567 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1569 2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
1571 * gst/rtsp-server/rtsp-client.c:
1572 * gst/rtsp-server/rtsp-client.h:
1573 rtsp-client: allow sub classes to adjust the seek
1574 Adds a new virtual function, adjust_play_mode(), that allows
1575 sub classes to adjust the seek done on the media. The sub class can
1576 modify the values of the the seek flags and the rate.
1577 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1579 2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
1581 * gst/rtsp-server/rtsp-media.c:
1582 * gst/rtsp-server/rtsp-media.h:
1583 * gst/rtsp-server/rtsp-stream.c:
1584 * gst/rtsp-server/rtsp-stream.h:
1585 * tests/check/gst/media.c:
1586 rtsp-media: allow specifying rate when seeking
1587 Add new function gst_rtsp_media_seek_full_with_rate() which allows the
1588 caller to specify the rate for the seek. Also added functions in
1589 rtsp-stream and rtsp-media for retreiving current rate and applied rate.
1590 https://bugzilla.gnome.org/show_bug.cgi?id=754575
1592 2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
1596 meson: Bump minimal GLib version to 2.44
1597 This means we can use some newer features and get rid of some
1598 boilerplate code using the G_DECLARE_* macros.
1599 As discussed on IRC, 2.44 is old enough by now to start depending on it.
1601 2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1603 * docs/libs/.gitignore:
1604 * docs/libs/Makefile.am:
1605 * docs/libs/gst-rtsp-server-docs.sgml:
1606 * docs/libs/gst-rtsp-server-sections.txt:
1607 * docs/libs/gst-rtsp-server.types:
1608 docs: remove obsolete gtk-doc related files
1610 2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1612 * gst/rtsp-sink/gstrtspclientsink.c:
1613 doc: remove xml from comments
1615 2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
1617 * docs/gst_plugins_cache.json:
1619 docs: Stop building the doc cache by default
1620 And update the cache
1621 Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
1623 2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
1625 * docs/gst_plugins_cache.json:
1626 docs: Update plugins documentation cache
1628 2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
1631 * gst/rtsp-server/rtsp-context.c:
1632 * gst/rtsp-server/rtsp-session-pool.c:
1633 doc: Fix some docstrings
1635 2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
1641 * docs/gst_plugins_cache.json:
1644 * docs/plugin-index.md:
1645 * docs/plugin-sitemap.txt:
1648 * docs/version.entities.in:
1649 * gst/rtsp-server/meson.build:
1650 * gst/rtsp-sink/meson.build:
1652 * meson_options.txt:
1653 docs: Port to hotdoc
1655 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1657 * gst/rtsp-server/rtsp-auth.c:
1658 * gst/rtsp-server/rtsp-client.h:
1659 rtsp-server: Fix various Since markers
1661 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
1663 * gst/rtsp-server/rtsp-media.c:
1664 * gst/rtsp-server/rtsp-sdp.c:
1665 * gst/rtsp-server/rtsp-session-media.c:
1666 * gst/rtsp-server/rtsp-stream.c:
1667 rtsp-server: Add various Since: 1.14 markers
1669 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1671 * gst/rtsp-server/rtsp-media-factory.c:
1672 * gst/rtsp-server/rtsp-media.c:
1673 * gst/rtsp-server/rtsp-stream-transport.c:
1674 * gst/rtsp-server/rtsp-stream.c:
1675 rtsp-server: Add various missing Since: 1.16 markers
1677 2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
1679 * gst/rtsp-sink/gstrtspclientsink.c:
1680 rtspclientsink: Set async-handling=false for the internal bins
1681 Without this we can easily run into a race condition with async state changes:
1682 - the pipeline is doing an async state change
1683 - we set the internal bins to PLAYING but that's ignored because an
1684 async state change is currently pending
1685 - the async state change finishes but does not change the state of the
1686 internal bins because of locked_state==TRUE
1687 - the internal bins stay in PAUSED forever
1689 2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1691 * gst/rtsp-sink/gstrtspclientsink.c:
1692 rtspclientsink: Use write_messages() API to send buffer lists in one go
1693 And to write messages with multiple memories also via writev().
1695 2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
1697 * gst/rtsp-server/rtsp-client.c:
1698 * gst/rtsp-server/rtsp-client.h:
1699 * gst/rtsp-server/rtsp-server-object.h:
1700 * gst/rtsp-server/rtsp-server.c:
1701 rtsp-client: Handle Content-Length limitation
1702 Add functionality to limit the Content-Length.
1703 API addition, Enhancement.
1704 Define an appropriate request size limit and reject requests
1705 exceeding the limit with response status 413 Request Entity Too Large
1708 2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1715 === release 1.16.0 ===
1717 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
1723 * gst-rtsp-server.doap:
1727 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
1729 * gst/rtsp-sink/gstrtspclientsink.c:
1730 rtspclientsink: Notify the stream transport about each written message
1731 Otherwise it will never try to send us the next one: it tries to keep
1732 exactly one message in-flight all the time.
1733 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
1734 in the client sink we always write data out synchronously.
1736 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
1738 * gst/rtsp-server/rtsp-stream.c:
1739 rtsp_server: Free thread pool before clean transport cache
1740 If not waiting for free thread pool before clean transport caches, there
1741 can be a crash if a thread is executing in transport list loop in
1742 function send_tcp_message.
1743 Also add a check if priv->send_pool in on_message_sent to avoid that a
1744 new thread is pushed during wait of free thread pool. This is possible
1745 since when waiting for free thread pool mutex have to be unlocked.
1747 === release 1.15.90 ===
1749 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
1755 * gst-rtsp-server.doap:
1759 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
1761 * gst/rtsp-server/rtsp-stream.c:
1762 rtsp-stream: Add support for GCM (RFC 7714)
1765 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
1767 * gst/rtsp-server/rtsp-session-pool.c:
1768 session pool: fix missing klass-> in klass->create_session
1770 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1773 g-i: pass --quiet to g-ir-scanner
1774 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
1775 that we get even if everything works just fine.
1776 We still get g-ir-scanner warnings and compiler warnings if
1777 we pass this option.
1779 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1782 g-i: silence 'nested extern' compiler warnings when building scanner binary
1783 We need a nested extern in our init section for the scanner binary
1784 so we can call gst_init to make sure GStreamer types are initialised
1785 (they are not all lazy init via get_type functions, but some are in
1786 exported variables). There doesn't seem to be any other mechanism to
1787 achieve this, so just remove that warning, it's not important at all.
1789 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1792 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
1794 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
1796 * gst/rtsp-server/rtsp-media.c:
1797 * tests/check/gst/media.c:
1798 rtsp-media: Handle set state when preparing.
1799 Handle the situation when a call to gst_rtsp_media_set_state is done
1800 when media status is preparing.
1801 Also add unit test for this scenario.
1802 The unit test simulate on a media level when two clients share a (live)
1804 Both clients have done SETUP and got responses. Now client 1 is doing
1805 play and client 2 is just closing the connection.
1806 Then without patch there are a problem when
1807 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
1808 And client2 is doing closing connection we can end up in a call
1809 to gst_rtsp_media_set_state when
1810 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
1811 shut down media is jumped over .
1812 With this patch and this scenario we wait until
1813 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
1814 execute after that and now we will execute the logic for
1817 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
1825 === release 1.15.2 ===
1827 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
1833 * gst-rtsp-server.doap:
1837 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
1839 * gst/rtsp-server/rtsp-media.c:
1840 * tests/check/gst/client.c:
1841 rtsp-media: Fix multicast use case with common media
1850 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
1852 * gst/rtsp-server/rtsp-client.c:
1853 * gst/rtsp-server/rtsp-stream.c:
1854 * gst/rtsp-server/rtsp-stream.h:
1855 rtsp-server: remove recursive behavior
1856 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
1858 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
1860 * gst/rtsp-server/rtsp-client.c:
1861 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
1862 And route all messages through the send_func if no send_messages_func
1864 We otherwise break backwards compatibility.
1866 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
1868 * docs/libs/gst-rtsp-server-sections.txt:
1869 * gst/rtsp-server/rtsp-client.c:
1870 * gst/rtsp-server/rtsp-client.h:
1871 * gst/rtsp-server/rtsp-stream.c:
1872 rtsp-client: Add support for sending buffer lists directly
1873 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
1875 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1877 * docs/libs/gst-rtsp-server-sections.txt:
1878 * gst/rtsp-server/rtsp-client.c:
1879 * gst/rtsp-server/rtsp-media.c:
1880 * gst/rtsp-server/rtsp-stream-transport.c:
1881 * gst/rtsp-server/rtsp-stream-transport.h:
1882 * gst/rtsp-server/rtsp-stream.c:
1883 * gst/rtsp-sink/gstrtspclientsink.c:
1884 rtsp-server: Add support for buffer lists
1885 This adds new functions for passing buffer lists through the different
1886 layers without breaking API/ABI, and enables the appsink to actually
1887 provide buffer lists.
1888 This should already reduce CPU usage and potentially context switches a
1889 bit by passing a whole buffer list from the appsink instead of
1890 individual buffers. As a next step it would be necessary to
1891 a) Add support for a vector of data for the GstRTSPMessage body
1892 b) Add support for sending multiple messages at once to the
1893 GstRTSPWatch and let it be handled internally
1894 c) Adding API to GOutputStream that works like writev()
1895 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
1897 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
1899 * gst/rtsp-server/rtsp-client.c:
1900 client: Fix crash in close handler
1901 The close handler could trigger a crash because it invalidated the
1902 watch_context while still leaving a source attached to it which would be
1903 cleaned up at a later point.
1905 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
1907 * gst/rtsp-server/rtsp-stream.c:
1908 rtsp-stream: Use cached address when allocating sockets
1909 If an address/port was previously decided upon (ex: multicast in the
1910 SDP), then use that instead of re-creating another one
1911 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
1913 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
1915 * gst/rtsp-server/rtsp-media.c:
1916 rtsp-media: Fix race codition in finish_unprepare
1917 The previous fix for race condition around finish_unprepare where the
1918 function could be called twice assumed that the status wouldn't change
1919 during execution of the function. This assumption is incorrect as the
1920 state may change, for example if an error message arrives from the
1922 Instead a flag keeping track on whether the finish_unprepare function
1923 is currently executing is introduced and checked.
1924 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
1926 === release 1.15.1 ===
1928 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1934 * gst-rtsp-server.doap:
1938 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
1940 * gst/rtsp-server/rtsp-stream.c:
1941 Add source elements to the pipeline before activation
1942 In plug_src we changed the element state before adding it to
1943 the owner container. This prevented the pipeline from intercepting
1944 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
1945 to assign a custom task pool.
1946 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
1948 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
1951 Automatic update of common submodule
1952 From ed78bee to 59cb678
1954 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
1956 * examples/test-appsrc.c:
1957 examples: test-appsrc: fix coding style error
1959 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
1961 * examples/test-appsrc.c:
1962 examples: test-appsrc: fix buffer leak
1964 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
1966 * gst/rtsp-server/rtsp-media.c:
1967 rtsp-media: Update priv->blocked when linked streams are unblocked.
1968 Media is considered to be blocked when all streams that belong to
1969 that media are blocked.
1970 This patch solves the problem of inconsistent updates of
1971 priv->blocked that are not synchronized with the media state.
1973 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
1975 * gst/rtsp-server/rtsp-media.c:
1976 rtsp-media: Don't block streams before seeking
1977 Before the seek operation is performed on media, it's required that
1978 its pipeline is prepared <=> the pipeline is in the PAUSED state.
1979 At this stage, all transport parts (transport sinks) have been successfully
1980 added to the pipeline and there is no need for blocking the streams.
1982 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
1984 * tests/check/gst/rtspserver.c:
1985 tests: rtspserver: Add shared media test case for TCP
1987 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
1989 * gst/rtsp-server/rtsp-stream.c:
1990 rtsp-stream: Use seqnum-offset for rtpinfo
1991 The sequence number in the rtpinfo is supposed to be the first RTP
1992 sequence number. The "seqnum" property on a payloader is supposed to be
1993 the number from the last processed RTP packet. The sequence number for
1994 payloaders that inherit gstrtpbasepayload will not be correct in case of
1995 buffer lists. In order to fix the seqnum property on the payloaders
1996 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
1997 "seqnum-offset" from the "stats" property contains the value of the
1998 very first RTP packet in a stream. The server will, however, try to look
1999 at the last simple in the sink element and only use properties on the
2000 payloader in case there no sink elements yet, and by looking at the last
2001 sample of the sink gives the server full control of which RTP packet it
2002 looks at. If the payloader does not have the "stats" property, "seqnum"
2003 is still used since "seqnum-offset" is only present in as part of
2004 "stats" and this is still an issue not solved with this patch.
2005 Needed for gst-plugins-base!17
2007 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
2009 * gst/rtsp-server/rtsp-stream.c:
2010 rtsp-stream: Plug memory leak
2011 Attaching a GSource to a context will increase the refcount. The idle
2012 source will never be free'd since the initial reference is never
2015 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
2018 Add Gitlab CI configuration
2019 This commit adds a .gitlab-ci.yml file, which uses a feature
2020 to fetch the config from a centralized repository. The intent is
2021 to have all the gstreamer modules use the same configuration.
2022 The configuration is currently hosted at the gst-ci repository
2023 under the gitlab/ci_template.yml path.
2024 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2026 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
2029 * gst-rtsp-server.doap:
2030 Update git locations to gitlab
2032 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
2034 * gst/rtsp-server/meson.build:
2035 meson: add new onvif types
2037 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
2039 * gst/rtsp-server/meson.build:
2040 Add ONVIF subclass headers to the installed headers in meson.build too
2042 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2044 * gst/rtsp-server/rtsp-server-object.h:
2045 * gst/rtsp-server/rtsp-server.h:
2046 rtsp-server: Declare GstRTSPServer struct before anything else
2047 It's needed by all kinds of other headers, including the ones that are
2048 required for defining the GstRTSPServer struct itself and its API.
2050 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2052 * gst/rtsp-server/rtsp-onvif-client.h:
2053 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2054 * gst/rtsp-server/rtsp-onvif-media.h:
2055 * gst/rtsp-server/rtsp-onvif-server.h:
2056 Mark all ONVIF-specific subclasses as Since 1.14
2058 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
2060 * gst/rtsp-server/Makefile.am:
2061 * gst/rtsp-server/meson.build:
2062 * gst/rtsp-server/rtsp-context.h:
2063 * gst/rtsp-server/rtsp-onvif-server.c:
2064 * gst/rtsp-server/rtsp-onvif-server.h:
2065 * gst/rtsp-server/rtsp-server-object.h:
2066 * gst/rtsp-server/rtsp-server-prelude.h:
2067 * gst/rtsp-server/rtsp-server.c:
2068 * gst/rtsp-server/rtsp-server.h:
2069 * gst/rtsp-server/rtsp-session.h:
2070 Include ONVIF types from single-include rtsp-server.h
2071 ... by actually making it a single-include header and moving everything
2072 related to the GstRTSPServer type to rtsp-server-object.h instead.
2073 Otherwise there are too many circular includes.
2074 https://bugzilla.gnome.org/show_bug.cgi?id=797361
2076 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
2078 * gst/rtsp-server/rtsp-client.c:
2079 * gst/rtsp-server/rtsp-latency-bin.c:
2080 * gst/rtsp-server/rtsp-stream.c:
2081 * gst/rtsp-server/rtsp-stream.h:
2082 rtsp-stream: use idle source in on_message_sent
2083 When the underlying layers are running on_message_sent, this sometimes
2084 causes the underlying layer to send more data, which will cause the
2085 underlying layer to run callback on_message_sent again. This can go on
2087 To break this chain, we introduce an idle source that takes care of
2088 sending data if there are more to send when running callback
2089 https://bugzilla.gnome.org/show_bug.cgi?id=797289
2091 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
2093 * gst/rtsp-server/rtsp-client.c:
2094 rtsp-client: Remove timeout GSource on cleanup
2095 Avoids ending up with races where a timeout would still be around
2096 *after* a client was gone. This could happen rather easily in
2097 RTSP-over-HTTP mode on a local connection, where each RTSP message
2098 would be sent as a different HTTP connection with the same tunnelid.
2099 If not properly removed, that timeout would then try to free again
2100 a client (and its contents).
2102 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2104 * gst/rtsp-server/Makefile.am:
2105 autotools: fix distcheck
2107 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2109 * gst/rtsp-server/Makefile.am:
2110 * gst/rtsp-server/meson.build:
2111 * gst/rtsp-server/rtsp-latency-bin.c:
2112 * gst/rtsp-server/rtsp-latency-bin.h:
2113 * gst/rtsp-server/rtsp-onvif-media.c:
2114 onvif: encapsulate onvif part into a bin
2115 ...and thus do not let onvif affect pipelines latency
2116 https://bugzilla.gnome.org/show_bug.cgi?id=797174
2118 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
2120 * tests/check/gst/client.c:
2121 tests: client: Avoid bind() failures in tests
2122 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2124 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
2126 * gst/rtsp-server/rtsp-media-factory.c:
2127 * gst/rtsp-server/rtsp-media-factory.h:
2128 * gst/rtsp-server/rtsp-media.c:
2129 * gst/rtsp-server/rtsp-media.h:
2130 * gst/rtsp-server/rtsp-stream.c:
2131 * gst/rtsp-server/rtsp-stream.h:
2132 * tests/check/gst/client.c:
2133 * tests/check/gst/mediafactory.c:
2134 New property for socket binding to mcast addresses
2135 By default the multicast sockets are bound to INADDR_ANY,
2136 as it's not allowed to bind sockets to multicast addresses
2137 in Windows. This default behaviour can be changed by setting
2138 bind-mcast-address property on the media-factory object.
2139 https://bugzilla.gnome.org/show_bug.cgi?id=797059
2141 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2144 * gst/rtsp-server/Makefile.am:
2145 * gst/rtsp-server/meson.build:
2146 * gst/rtsp-server/rtsp-address-pool.c:
2147 * gst/rtsp-server/rtsp-auth.c:
2148 * gst/rtsp-server/rtsp-client.c:
2149 * gst/rtsp-server/rtsp-context.c:
2150 * gst/rtsp-server/rtsp-media-factory-uri.c:
2151 * gst/rtsp-server/rtsp-media-factory.c:
2152 * gst/rtsp-server/rtsp-media.c:
2153 * gst/rtsp-server/rtsp-mount-points.c:
2154 * gst/rtsp-server/rtsp-params.c:
2155 * gst/rtsp-server/rtsp-permissions.c:
2156 * gst/rtsp-server/rtsp-sdp.c:
2157 * gst/rtsp-server/rtsp-server-prelude.h:
2158 * gst/rtsp-server/rtsp-server.c:
2159 * gst/rtsp-server/rtsp-session-media.c:
2160 * gst/rtsp-server/rtsp-session-pool.c:
2161 * gst/rtsp-server/rtsp-session.c:
2162 * gst/rtsp-server/rtsp-stream-transport.c:
2163 * gst/rtsp-server/rtsp-stream.c:
2164 * gst/rtsp-server/rtsp-thread-pool.c:
2165 * gst/rtsp-server/rtsp-token.c:
2167 libs: fix API export/import and 'inconsistent linkage' on MSVC
2168 Export rtsp-server library API in headers when we're building the
2169 library itself, otherwise import the API from the headers.
2170 This fixes linker warnings on Windows when building with MSVC.
2171 Fix up some missing config.h includes when building the lib which
2172 is needed to get the export api define from config.h
2173 https://bugzilla.gnome.org/show_bug.cgi?id=797185
2175 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
2177 * gst/rtsp-server/rtsp-media-factory.c:
2178 rtsp-media-factory: Add missing break statements
2179 This resulted in warnings/assertions whenever one accessed the
2180 max-mcast-ttl property.
2184 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
2187 * meson_options.txt:
2188 meson: add gobject-cast-checks, glib-asserts, glib-checks options
2190 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
2193 * meson_options.txt:
2194 * tests/check/meson.build:
2195 meson: add option to disable build of rtspclientsink plugin
2197 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2199 * meson_options.txt:
2200 meson: re-arrange options
2202 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2205 * meson_options.txt:
2206 * tests/check/meson.build:
2207 * tests/meson.build:
2208 meson: Use feature option for tests option
2209 This was somehow missed the last time around.
2211 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2213 * gst/rtsp-server/meson.build:
2215 meson: Maintain macOS ABI through dylib versioning
2216 Requires Meson 0.48, but the feature will be ignored on older versions
2217 so it's safe to add it without bumping the requirement.
2219 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2221 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
2223 * gst/rtsp-sink/meson.build:
2225 meson: add pkg-config file for the rtspclientsink plugin
2227 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2229 * gst/rtsp-server/rtsp-client.c:
2230 * tests/check/gst/client.c:
2231 rtsp-client: Avoid reuse of channel numbers for interleaved
2232 If a (strange) client would reuse interleaved channel numbers in
2233 multiple SETUP requests, we should not accept them. The channel
2234 numbers are used for looking up stream transports in the
2235 priv->transports hash table, and transports disappear from the table
2236 if channel numbers are reused.
2237 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
2238 server to change the channel numbers suggested by the client.
2239 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2241 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
2243 * tests/check/gst/client.c:
2244 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
2245 Allow regex for matching transport header against expected pattern.
2246 https://bugzilla.gnome.org/show_bug.cgi?id=796988
2248 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2250 * tests/check/meson.build:
2251 meson: There is no gstreamer-plugins-good-1.0.pc
2252 There is no installed version of that, only an uninstalled version.
2254 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
2256 * gst/rtsp-server/rtsp-client.c:
2257 * tests/check/gst/stream.c:
2258 Fix indentation again
2260 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
2262 * gst/rtsp-server/rtsp-client.c:
2263 * gst/rtsp-server/rtsp-stream.c:
2264 * gst/rtsp-server/rtsp-stream.h:
2265 * tests/check/gst/client.c:
2266 * tests/check/gst/stream.c:
2267 stream: Added a list of multicast client addresses
2268 When media is shared, the same media stream can be sent
2269 to multiple multicast groups. Currently, there is no API
2270 to retrieve multicast addresses from the stream.
2271 When calling gst_rtsp_stream_get_multicast_address() function,
2272 only the first multicast address is returned.
2273 With this patch, each multicast destination requested in SETUP
2274 will be stored in an internal list (call to
2275 gst_rtsp_stream_add_multicast_client_address()).
2276 The list of multicast groups requested by the clients can be
2277 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
2278 There still exist some problems with the current implementation
2279 in the multicast case:
2280 1) The receiving part is currently only configured with
2281 regard to the first multicast client (see
2282 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2283 2) Secondly, of security reasons, some constraints should be
2284 put on the requested multicast destinations (see
2285 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
2286 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
2287 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2289 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2291 * gst/rtsp-server/rtsp-client.c:
2292 * gst/rtsp-server/rtsp-stream.c:
2293 * gst/rtsp-server/rtsp-stream.h:
2294 * tests/check/gst/client.c:
2295 stream: Choose the maximum ttl value provided by multicast clients
2296 The maximum ttl value provided so far by the multicast clients
2297 will be chosen and reported in the response to the current
2299 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
2300 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2302 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2304 * gst/rtsp-server/rtsp-stream.c:
2305 * tests/check/gst/client.c:
2306 rtsp-stream: Don't require address pool in the transport specific case
2307 If "transport.client-settings" parameter is set to true, the client is
2308 allowed to specify destination, ports and ttl.
2309 There is no need for pre-configured address pool.
2310 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
2311 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2313 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
2315 * gst/rtsp-server/rtsp-client.c:
2316 * tests/check/gst/client.c:
2317 client: Don't reserve multicast address in the client setting case
2318 When two multicast clients request specific transport
2319 configurations, and "transport.client-settings" parameter is
2320 set to true, it's wrong to actually require that these two
2321 clients request the same multicast group.
2322 Removed test_client_multicast_invalid_transport_specific test
2323 cases as they wrongly require that the requested destination
2324 address is supposed to be present in the address pool, also in
2325 the case when "transport.client-settings" parameter is set to true.
2326 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
2327 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2329 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2331 * gst/rtsp-server/rtsp-media-factory.c:
2332 * gst/rtsp-server/rtsp-media-factory.h:
2333 * gst/rtsp-server/rtsp-media.c:
2334 * gst/rtsp-server/rtsp-media.h:
2335 * gst/rtsp-server/rtsp-stream.c:
2336 * gst/rtsp-server/rtsp-stream.h:
2337 * tests/check/gst/mediafactory.c:
2338 Add new API for setting/getting maximum multicast ttl value
2339 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
2340 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2342 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2344 * gst/rtsp-server/rtsp-stream.c:
2345 rtsp-stream: avoid duplicating the first multicast client
2346 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2347 clients were dynamically added and removed to the multicast
2348 udp sinks, as such we should no longer add a first client in
2349 set_multicast_socket_for_udpsink
2350 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2352 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
2354 * gst/rtsp-server/rtsp-stream.c:
2355 Revert "rtsp-stream: avoid duplicating the first multicast client"
2356 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
2357 Commits where accidentially squashed together
2359 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2361 * gst/rtsp-server/rtsp-client.c:
2362 * gst/rtsp-server/rtsp-media-factory.c:
2363 * gst/rtsp-server/rtsp-media-factory.h:
2364 * gst/rtsp-server/rtsp-media.c:
2365 * gst/rtsp-server/rtsp-media.h:
2366 * gst/rtsp-server/rtsp-stream.c:
2367 * gst/rtsp-server/rtsp-stream.h:
2368 * tests/check/gst/client.c:
2369 * tests/check/gst/mediafactory.c:
2370 Revert "Add new API for setting/getting maximum multicast ttl value"
2371 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
2372 Commits where accidentially squashed together
2374 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
2376 * gst/rtsp-server/rtsp-stream.c:
2377 * tests/check/gst/client.c:
2378 Revert "rtsp-stream: Don't require address pool in the transport specific case"
2379 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
2380 Commits where accidentially squashed together
2382 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2384 * gst/rtsp-server/rtsp-client.c:
2385 * gst/rtsp-server/rtsp-stream.c:
2386 * gst/rtsp-server/rtsp-stream.h:
2387 * tests/check/gst/client.c:
2388 * tests/check/gst/stream.c:
2389 Revert "stream: Choose the maximum ttl value provided by multicast clients"
2390 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
2391 Commits where accidentially squashed together
2393 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
2395 * examples/test-auth-digest.c:
2396 examples: Fix indentation
2398 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
2400 * gst/rtsp-server/rtsp-client.c:
2401 * gst/rtsp-server/rtsp-stream.c:
2402 * gst/rtsp-server/rtsp-stream.h:
2403 * tests/check/gst/client.c:
2404 * tests/check/gst/stream.c:
2405 stream: Choose the maximum ttl value provided by multicast clients
2406 The maximum ttl value provided so far by the multicast clients
2407 will be chosen and reported in the response to the current
2409 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2411 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
2413 * gst/rtsp-server/rtsp-stream.c:
2414 * tests/check/gst/client.c:
2415 rtsp-stream: Don't require address pool in the transport specific case
2416 If "transport.client-settings" parameter is set to true, the client is
2417 allowed to specify destination, ports and ttl.
2418 There is no need for pre-configured address pool.
2419 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2421 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
2423 * gst/rtsp-server/rtsp-client.c:
2424 * gst/rtsp-server/rtsp-media-factory.c:
2425 * gst/rtsp-server/rtsp-media-factory.h:
2426 * gst/rtsp-server/rtsp-media.c:
2427 * gst/rtsp-server/rtsp-media.h:
2428 * gst/rtsp-server/rtsp-stream.c:
2429 * gst/rtsp-server/rtsp-stream.h:
2430 * tests/check/gst/client.c:
2431 * tests/check/gst/mediafactory.c:
2432 Add new API for setting/getting maximum multicast ttl value
2433 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2435 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2437 * gst/rtsp-server/rtsp-stream.c:
2438 rtsp-stream: avoid duplicating the first multicast client
2439 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
2440 clients were dynamically added and removed to the multicast
2441 udp sinks, as such we should no longer add a first client in
2442 set_multicast_socket_for_udpsink
2443 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2445 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
2447 * gst/rtsp-server/Makefile.am:
2448 rtsp-server: Add gstreamer-base gir dir in autotools
2450 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2452 * gst/rtsp-server/rtsp-client.c:
2453 * gst/rtsp-server/rtsp-stream.c:
2454 rtsp-client: always allocate both IPV4 and IPV6 sockets
2455 multiudpsink does not support setting the socket* properties
2456 after it has started, which meant that rtsp-server could no
2457 longer serve on both IPV4 and IPV6 sockets since the patches
2458 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
2460 When first connecting an IPV6 client then an IPV4 client,
2461 multiudpsink fell back to using the IPV6 socket.
2462 When first connecting an IPV4 client, then an IPV6 client,
2463 multiudpsink errored out, released the IPV4 socket, then
2464 crashed when trying to send a message on NULL nevertheless,
2465 that is however a separate issue.
2466 This could probably be fixed by handling the setting of
2467 sockets in multiudpsink after it has started, that will
2468 however be a much more significant effort.
2469 For now, this commit simply partially reverts the behaviour
2470 of rtsp-stream: it will continue to only create the udpsinks
2471 when needed, as was the case since the patches were merged,
2472 it will however when creating them, always allocate both
2473 sockets and set them on the sink before it starts, as was
2474 the case prior to the patches.
2475 Transport configuration will only error out if the allocation
2476 of UDP sockets fails for the actual client's family, this
2477 also downgrades the GST_ERRORs in alloc_ports_one_family
2478 to GST_WARNINGs, as failing to allocate is no longer
2480 https://bugzilla.gnome.org/show_bug.cgi?id=796875
2482 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2485 * meson_options.txt:
2486 meson: Convert common options to feature options
2487 These are necessary for gst-build to set options correctly. The
2488 remaining automagic option is cgroup support in examples.
2489 https://bugzilla.gnome.org/show_bug.cgi?id=795107
2491 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2493 * gst/rtsp-server/rtsp-stream.c:
2494 rtsp-stream: Slightly simplify locking
2496 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
2498 * gst/rtsp-server/rtsp-client.c:
2499 * gst/rtsp-server/rtsp-stream-transport.c:
2500 * gst/rtsp-server/rtsp-stream-transport.h:
2501 * gst/rtsp-server/rtsp-stream.c:
2502 Limit queued TCP data messages to one per stream
2503 Before, the watch backlog size in GstRTSPClient was changed
2504 dynamically between unlimited and a fixed size, trying to avoid both
2505 unlimited memory usage and deadlocks while waiting for place in the
2506 queue. (Some of the deadlocks were described in a long comment in
2508 In the previous commit, we changed to a fixed backlog size of 100.
2509 This is possible, because we now handle RTP/RTCP data messages differently
2510 from RTSP request/response messages.
2511 The data messages are messages tunneled over TCP. We allow at most one
2512 queued data message per stream in GstRTSPClient at a time, and
2513 successfully sent data messages are acked by sending a "message-sent"
2514 callback from the GstStreamTransport. Until that ack comes, the
2515 GstRTSPStream does not call pull_sample() on its appsink, and
2516 therefore the streaming thread in the pipeline will not be blocked
2517 inside GstRTSPClient, waiting for a place in the queue.
2518 pull_sample() is called when we have both an ack and a "new-sample"
2519 signal from the appsink. Then, we know there is a buffer to write.
2520 RTSP request/response messages are not acked in the same way as data
2521 messages. The rest of the 100 places in the queue are used for
2522 them. If the queue becomes full of request/response messages, we
2523 return an error and close the connection to the client.
2524 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2526 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
2528 * gst/rtsp-server/rtsp-client.c:
2529 rtsp-client: Use fixed backlog size
2530 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
2531 Preparation for the next commit, which changes to a different way of
2532 avoiding both deadlocks and unlimited memory usage with the watch
2535 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2537 * gst/rtsp-server/rtsp-media.c:
2538 rtsp-media: unref clock (if set) when finalizing
2539 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2541 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
2543 * docs/libs/gst-rtsp-server-sections.txt:
2544 rtsp-media: add gst_rtsp_media_*_set_clock to docs
2545 https://bugzilla.gnome.org/show_bug.cgi?id=796814
2547 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
2549 * gst/rtsp-server/rtsp-media-factory.c:
2550 media-factory: unref old clock when setting new clock
2551 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2553 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
2555 * gst/rtsp-server/rtsp-media-factory.c:
2556 media-factory: unref clock in finalize
2557 https://bugzilla.gnome.org/show_bug.cgi?id=796724
2559 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
2561 * gst/rtsp-server/rtsp-onvif-media.c:
2562 rtsp-onvif-media: fix g-ir-scanner warnings
2564 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2567 .gitignore: add another example binary
2569 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
2571 * examples/meson.build:
2572 meson: add new test-appsrc2 example to meson build
2574 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2576 * examples/Makefile.am:
2577 examples: fix build of new test-appsrc2 example
2578 Need to link against libgstapp-1.0.
2580 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
2582 * examples/.gitignore:
2583 * examples/Makefile.am:
2584 * examples/test-appsrc2.c:
2585 examples: Add test-appsrc2
2586 Add an example of feeding both audio and video into an RTSP
2587 pipeline via appsrc.
2589 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
2591 * gst/rtsp-server/rtsp-client.c:
2592 client: Strip transport parts as whitespaces could be around commas
2593 https://bugzilla.gnome.org/show_bug.cgi?id=758428
2595 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
2597 * gst/rtsp-server/rtsp-stream.c:
2598 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
2599 Fix race when setting up source elements.
2600 Since we set the source element(s) to PLAYING state before hooking
2601 them up to the downstream funnel, it's possible for the source element
2602 to receive packets before we actually get to linking it to the funnel,
2603 in which case buffers would be pushed out on an unlinked pad, causing
2604 it to error out and stop receiving more data.
2605 We fix this by blocking the source's srcpad until we have linked it.
2606 https://bugzilla.gnome.org/show_bug.cgi?id=796160
2608 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
2610 * gst/rtsp-server/rtsp-stream.c:
2611 rtsp-stream: Fix mismatch between allowed and configured protocols
2612 https://bugzilla.gnome.org/show_bug.cgi?id=796679
2614 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
2616 * gst/rtsp-server/rtsp-stream.c:
2617 rtsp-stream: Emit a signal when the SRTP decoder is created
2618 https://bugzilla.gnome.org/show_bug.cgi?id=778080
2620 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
2622 * gst/rtsp-server/rtsp-stream.c:
2623 rtsp-stream: Don't require presence of sinks in _get_*_socket()
2624 Transport specific sink elements are added to the pipeline
2625 in PLAY request and sockets are already created in SETUP so
2626 it's actually wrong to require the presence of sinks in
2627 _get_*_socket() functions.
2628 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2630 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
2632 * gst/rtsp-server/rtsp-stream.c:
2633 rtsp-stream: Update transport for multicast clients as well
2634 If a multicast client requests different transport settings
2635 than the existing one make sure that this new transport
2636 configuruation is propagated to the multicast udp sink.
2637 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2639 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
2641 * gst/rtsp-server/rtsp-stream.c:
2642 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
2643 And not on unicast udp sinks
2644 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2646 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
2648 * gst/rtsp-server/rtsp-address-pool.c:
2649 * gst/rtsp-server/rtsp-auth.c:
2650 * gst/rtsp-server/rtsp-client.c:
2651 * gst/rtsp-server/rtsp-media-factory-uri.c:
2652 * gst/rtsp-server/rtsp-media-factory.c:
2653 * gst/rtsp-server/rtsp-media.c:
2654 * gst/rtsp-server/rtsp-mount-points.c:
2655 * gst/rtsp-server/rtsp-server.c:
2656 * gst/rtsp-server/rtsp-session-media.c:
2657 * gst/rtsp-server/rtsp-session-pool.c:
2658 * gst/rtsp-server/rtsp-session.c:
2659 * gst/rtsp-server/rtsp-stream-transport.c:
2660 * gst/rtsp-server/rtsp-stream.c:
2661 * gst/rtsp-server/rtsp-thread-pool.c:
2662 Update for g_type_class_add_private() deprecation in recent GLib
2664 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
2666 * gst/rtsp-server/rtsp-auth.c:
2667 * gst/rtsp-server/rtsp-media.c:
2668 * gst/rtsp-server/rtsp-sdp.c:
2669 * gst/rtsp-server/rtsp-stream.c:
2672 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
2674 * examples/Makefile.am:
2675 * examples/test-video-disconnect.c:
2676 examples: Add test-video-disconnect example
2677 Simple example which cuts off all clients 10 seconds
2678 after the first one connects.
2680 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2682 * docs/libs/gst-rtsp-server-sections.txt:
2683 * examples/test-auth-digest.c:
2684 * gst/rtsp-server/rtsp-auth.c:
2685 * gst/rtsp-server/rtsp-auth.h:
2686 rtsp-auth: Add support for parsing .htdigest files
2687 Passwords are usually not stored in clear text, but instead
2688 stored already hashed in a .htdigest file.
2689 Add support for parsing such files, add API to allow setting
2690 a custom realm in RTSPAuth, and update the digest example.
2691 https://bugzilla.gnome.org/show_bug.cgi?id=796637
2693 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
2695 * gst/rtsp-sink/gstrtspclientsink.c:
2696 * gst/rtsp-sink/gstrtspclientsink.h:
2697 rtspclientsink: fix waiting for multiple streams
2698 We were previously only ever waiting for a single stream to notify it's
2699 blocked status through GstRTSPStreamBlocking. Actually count streams to
2701 Fixes rtspclientsink sending SDP's without out some of the input
2703 https://bugzilla.gnome.org/show_bug.cgi?id=796624
2705 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2707 * docs/libs/gst-rtsp-server-sections.txt:
2708 docs: add missing auth methods
2710 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2712 * gst/rtsp-server/rtsp-stream.c:
2713 rtsp-stream: only create funnel if it didn't exist already.
2714 This precented using multiple protocols for the same stream.
2715 https://bugzilla.gnome.org/show_bug.cgi?id=796634
2717 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2719 * examples/meson.build:
2720 meson: build auth-digest example
2722 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
2724 * gst/rtsp-server/rtsp-client.c:
2725 * gst/rtsp-server/rtsp-media.c:
2726 * gst/rtsp-server/rtsp-sdp.c:
2727 * gst/rtsp-server/rtsp-session-media.c:
2728 * gst/rtsp-server/rtsp-stream-transport.c:
2729 Get payloader stats only for the sending streams
2730 Get/set payloader properties only for streams that actually
2731 contain a payloader element.
2732 https://bugzilla.gnome.org/show_bug.cgi?id=796523
2734 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
2736 * gst/rtsp-server/Makefile.am:
2737 Makefile: Don't hardcode libtool for g-i build
2738 Similar to the other commits in core/base/bad
2740 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
2742 * gst/rtsp-server/rtsp-onvif-media-factory.h:
2743 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
2744 https://bugzilla.gnome.org/show_bug.cgi?id=796229
2746 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
2748 * gst/rtsp-sink/gstrtspclientsink.c:
2749 rtspclientsink: Don't deadlock in preroll on early close
2750 If the connection is closed very early, the flushing
2751 marker might not get set and rtspclientsink can get
2752 deadlocked waiting for preroll forever.
2753 https://bugzilla.gnome.org/show_bug.cgi?id=786961
2755 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2758 * meson_options.txt:
2759 meson: Update option names to omit disable_ and with- prefixes
2760 Also yield common options to the outer project (gst-build in our case)
2761 so that they don't have to be set manually.
2763 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2766 meson: use -Wl,-Bsymbolic-functions where supported
2767 Just like the autotools build.
2769 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2772 * tests/check/Makefile.am:
2773 configure: check for -good and -bad plugins only in uninstalled setup
2774 Avoids confusing configure messages looking or a -good .pc file
2776 Also use plugindir variables that common macros set while at it.
2777 https://bugzilla.gnome.org/show_bug.cgi?id=795466
2779 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
2781 * gst/rtsp-server/rtsp-client.c:
2782 rtsp-client: Fix session timeout
2783 When streaming data over TCP then is not the keep-alive
2784 functionality working.
2785 The reason is that the function do_send_data have changed
2786 to boolean but the code is still checking the received result
2787 from send_func with GST_RTSP_OK.
2788 The result is that a successful send_func will always lead to
2789 that do_send_data is returning false and the keep-alive will
2791 https://bugzilla.gnome.org/show_bug.cgi?id=795321
2793 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2795 * docs/libs/gst-rtsp-server-sections.txt:
2796 * gst/rtsp-server/rtsp-media.c:
2797 * gst/rtsp-server/rtsp-sdp.c:
2798 * gst/rtsp-server/rtsp-stream.c:
2799 * gst/rtsp-server/rtsp-stream.h:
2800 * gst/rtsp-sink/gstrtspclientsink.c:
2801 * gst/rtsp-sink/gstrtspclientsink.h:
2802 Implement support for ULP Forward Error Correction
2803 In this initial commit, interface is only exposed for RECORD,
2804 further work will be needed in rtspsrc to support this for
2806 https://bugzilla.gnome.org/show_bug.cgi?id=794911
2808 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
2810 * gst/rtsp-server/rtsp-onvif-media.c:
2811 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
2812 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
2813 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
2814 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
2815 the opposite, just like the ONVIF standard.
2816 Let's follow those RFCs as we're doing RTSP here, and add a property at
2817 a later time if needed to switch to the SDP RFC behaviour.
2818 https://bugzilla.gnome.org/show_bug.cgi?id=793964
2820 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2823 Automatic update of common submodule
2824 From 3fa2c9e to ed78bee
2826 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
2828 * gst/rtsp-server/rtsp-client.c:
2829 * gst/rtsp-server/rtsp-media-factory.c:
2830 * gst/rtsp-server/rtsp-media.c:
2831 * gst/rtsp-server/rtsp-stream.c:
2832 * tests/check/gst/rtspclientsink.c:
2833 gst: Run everything through gst-indent again
2835 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
2837 * gst/rtsp-server/rtsp-media.c:
2838 * tests/check/gst/media.c:
2839 rtsp-media: query the position on active streams if media is complete
2840 If the media is complete, i.e. one or more streams have been configured
2841 with sinks, then we want to query the position on those streams only.
2842 A query on an incomplete stream may return a position that originates from
2844 https://bugzilla.gnome.org/show_bug.cgi?id=794964
2846 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2848 * gst/rtsp-sink/gstrtspclientsink.c:
2849 rtspclientsink: make sure not to use freed string
2850 Set transport string to NULL after freeing it, so that
2851 at worst we get a NULL pointer if constructing a new
2852 transport string fails (which shouldn't really fail here).
2853 Also check return value of that, just in case.
2856 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2858 * gst/rtsp-server/rtsp-client.c:
2859 rtsp-client: do not free string passed to take_header
2861 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2863 * gst/rtsp-server/rtsp-stream.c:
2864 rtsp-stream: do not take lock in request_aux_receiver
2865 Added it right before pushing the previous commit, it is
2866 incorrect and deadlocks because this function gets called
2867 from the join_bin thread, which already holds the lock,
2868 that's the reason why request_aux_sender didn't take the
2871 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2873 * docs/libs/gst-rtsp-server-sections.txt:
2874 * gst/rtsp-server/rtsp-media-factory.c:
2875 * gst/rtsp-server/rtsp-media-factory.h:
2876 * gst/rtsp-server/rtsp-media.c:
2877 * gst/rtsp-server/rtsp-media.h:
2878 * gst/rtsp-server/rtsp-stream.c:
2879 * gst/rtsp-server/rtsp-stream.h:
2880 rtsp-server: add API to enable retransmission requests
2881 "do-retransmission" was previously set when rtx-time != 0,
2882 which made no sense as do-retransmission is used to enable
2883 the sending of retransmission requests, where as rtx-time
2884 is used by the peer to enable storing of buffers in order
2885 to respond to retransmission requests.
2886 rtsp-media now also provides a callback for the
2887 request-aux-receiver signal.
2888 https://bugzilla.gnome.org/show_bug.cgi?id=794822
2890 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2892 * gst/rtsp-sink/gstrtspclientsink.c:
2893 rtspclientsink: add rtx ssrc to mikey's crypto sessions
2894 https://bugzilla.gnome.org/show_bug.cgi?id=794813
2896 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2898 * gst/rtsp-sink/gstrtspclientsink.c:
2899 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
2900 This in order to be able to decrypt the RTCP backchannel
2901 https://bugzilla.gnome.org/show_bug.cgi?id=794813
2903 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2905 * gst/rtsp-server/rtsp-client.c:
2906 rtsp-client: Send KeyMgmt header in ANNOUNCE response
2907 When sending back an encrypted RTCP back channel, it is useful
2908 for the client to know the encryption key.
2909 https://bugzilla.gnome.org/show_bug.cgi?id=794813
2911 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2913 * gst/rtsp-server/rtsp-client.c:
2914 * gst/rtsp-server/rtsp-stream.c:
2915 * gst/rtsp-server/rtsp-stream.h:
2916 rtsp-stream: extract handle_keymgmt from rtsp-client
2917 rtspclientsink will also need to parse KeyMgmt headers
2918 sent by the server to decrypt the RTCP backchannel stream
2919 https://bugzilla.gnome.org/show_bug.cgi?id=794813
2921 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2923 * gst/rtsp-sink/gstrtspclientsink.c:
2924 * tests/check/gst/rtspclientsink.c:
2925 rtspclientsink: Fix client ports for the RTCP backchannel
2926 This was broken since the work for delayed transport creation
2927 was merged: the creation of the transports string depends on
2928 calling stream_get_server_port, which only starts returning
2929 something meaningful after a call to stream_allocate_udp_sockets
2930 has been made, this function expects a transport that we parse
2931 from the transport string ...
2932 Significant refactoring is in order, but does not look entirely
2933 trivial, for now we put a band aid on and create a second transport
2934 string after the stream has been completed, to pass it in
2935 the request headers instead of the previous, incomplete one.
2936 https://bugzilla.gnome.org/show_bug.cgi?id=794789
2938 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
2940 * gst/rtsp-server/rtsp-client.c:
2941 rtsp-client:Error handling when equal http session cookie
2942 There are some clients that are sending same session cookie on random
2944 https://bugzilla.gnome.org/show_bug.cgi?id=753616
2946 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
2948 * gst/rtsp-server/rtsp-media-factory-uri.c:
2949 rtsp-media-factory-uri: Fix compilation with latest GLib
2950 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
2951 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
2952 data->factory = g_object_ref (factory);
2955 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2963 === release 1.14.0 ===
2965 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2971 * gst-rtsp-server.doap:
2975 === release 1.13.91 ===
2977 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
2983 * gst-rtsp-server.doap:
2987 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
2989 * gst/rtsp-server/Makefile.am:
2990 * gst/rtsp-server/meson.build:
2991 * gst/rtsp-server/rtsp-address-pool.h:
2992 * gst/rtsp-server/rtsp-auth.h:
2993 * gst/rtsp-server/rtsp-client.h:
2994 * gst/rtsp-server/rtsp-context.h:
2995 * gst/rtsp-server/rtsp-media-factory-uri.h:
2996 * gst/rtsp-server/rtsp-media-factory.h:
2997 * gst/rtsp-server/rtsp-media.h:
2998 * gst/rtsp-server/rtsp-mount-points.h:
2999 * gst/rtsp-server/rtsp-onvif-client.h:
3000 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3001 * gst/rtsp-server/rtsp-onvif-media.h:
3002 * gst/rtsp-server/rtsp-onvif-server.h:
3003 * gst/rtsp-server/rtsp-params.h:
3004 * gst/rtsp-server/rtsp-permissions.h:
3005 * gst/rtsp-server/rtsp-sdp.h:
3006 * gst/rtsp-server/rtsp-server-prelude.h:
3007 * gst/rtsp-server/rtsp-server.h:
3008 * gst/rtsp-server/rtsp-session-media.h:
3009 * gst/rtsp-server/rtsp-session-pool.h:
3010 * gst/rtsp-server/rtsp-session.h:
3011 * gst/rtsp-server/rtsp-stream-transport.h:
3012 * gst/rtsp-server/rtsp-stream.h:
3013 * gst/rtsp-server/rtsp-thread-pool.h:
3014 * gst/rtsp-server/rtsp-token.h:
3015 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
3016 We need different export decorators for the different libs.
3017 For now no actual change though, just rename before the release,
3018 and add prelude headers to define the new decorator to GST_EXPORT.
3020 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3022 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3023 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
3024 https://bugzilla.gnome.org/show_bug.cgi?id=794143
3026 === release 1.13.90 ===
3028 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
3034 * gst-rtsp-server.doap:
3038 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3040 * gst/rtsp-server/rtsp-media-factory.c:
3041 * gst/rtsp-server/rtsp-permissions.c:
3042 permissions: add Since tags and example for new API
3044 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3046 * docs/libs/gst-rtsp-server-sections.txt:
3047 * gst/rtsp-server/rtsp-media-factory.c:
3048 * gst/rtsp-server/rtsp-media-factory.h:
3049 * gst/rtsp-server/rtsp-permissions.c:
3050 * gst/rtsp-server/rtsp-permissions.h:
3051 * tests/check/gst/permissions.c:
3052 permissions: more bindings-friendly API
3053 https://bugzilla.gnome.org/show_bug.cgi?id=793975
3055 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3058 meson: enable more warnings
3060 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3062 * gst/rtsp-server/rtsp-client.c:
3063 rtsp-client: Place netaddress meta on packets received via TCP
3064 This allows us to later map signals from rtpbin/rtpsource back to the
3065 corresponding stream transport, and allows to do keep-alive based on
3066 RTCP packets in case of TCP media transport.
3067 https://bugzilla.gnome.org/show_bug.cgi?id=789646
3069 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3071 * gst/rtsp-sink/gstrtspclientsink.c:
3072 rtspclientsink: if OPEN failed, unqueue next command
3073 As READY_TO_PAUSED can no longer return async, the RECORD
3074 command will be queued before the OPEN command fails
3075 (for example in case the server could not be connected),
3076 and record then waits for ever.
3077 https://bugzilla.gnome.org/show_bug.cgi?id=793896
3079 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3081 * gst/rtsp-sink/gstrtspclientsink.c:
3082 rtspclientsink: fix retrieval of custom payloader caps
3083 If a bin is passed as the custom payloader, the caps of
3084 its factory will be empty, the correct way to obtain the caps
3085 is to query its sinkpad.
3087 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3089 * gst/rtsp-sink/gstrtspclientsink.c:
3090 rtspclientsink: fix extra unref of custom payloader
3092 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3094 * gst/rtsp-sink/gstrtspclientsink.c:
3095 rspclientsink: fix recent code indentation
3097 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3099 * gst/rtsp-sink/gstrtspclientsink.c:
3100 rtspclientsink: add missing get_type prototype
3102 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3104 * gst/rtsp-sink/gstrtspclientsink.c:
3105 rtspclientsink: allow setting payloader as pad property
3106 This was a FIXME item, and can be quite useful, also
3107 allowing to specify payloader properties from the command
3108 line, which is always nice.
3109 https://bugzilla.gnome.org/show_bug.cgi?id=793776
3111 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
3113 * gst/rtsp-server/rtsp-media.c:
3114 rtsp-media: Replace g_print() log line
3115 https://bugzilla.gnome.org/show_bug.cgi?id=793838
3117 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3119 * gst/rtsp-server/rtsp-media.c:
3120 * tests/check/gst/rtspclientsink.c:
3121 rtsp-media: fix RECORD getting stuck
3122 The test_record case was working because async=false had
3123 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
3124 but that was incorrect, as it should not be needed.
3125 Removing async=false made the test fail as expected, this is
3126 fixed by not trying to preroll when preparing the media for
3127 RECORD, as start_prepare is called upon receiving ANNOUNCE,
3128 and our peer will not start sending media until it has received
3129 a response to that request, and sent and received a response
3130 to RECORD as well, thus obviously preventing preroll.
3131 https://bugzilla.gnome.org/show_bug.cgi?id=793738
3133 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3135 * gst/rtsp-server/rtsp-auth.c:
3136 rtsp-auth: fix set_tls_authentication_mode annotation
3138 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
3140 * gst/rtsp-server/rtsp-onvif-media.c:
3141 rtp-server: remove redefined variable
3142 res is a boolean variable which is defined in the function scope and
3143 redefined, with no reason, in the loop scope. This patch removes the
3145 https://bugzilla.gnome.org/show_bug.cgi?id=793592
3147 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
3149 * gst/rtsp-server/rtsp-media.c:
3150 * gst/rtsp-server/rtsp-stream.c:
3151 * gst/rtsp-server/rtsp-stream.h:
3152 stream: Add functions for checking if stream is receiver or sender
3153 ...and replace all checks for RECORD in GstRTSPMedia which are really
3154 for "sender-only". This way the code becomes more generic and introducing
3155 support for onvif-backchannel later on will require no changes in
3158 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
3160 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3161 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3162 onvif: Make requires_backchannel() public
3163 ...in order to let subclasses building the onvif part of the pipeline
3164 check whether backchannel shall be included or not.
3166 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
3168 * gst/rtsp-server/rtsp-onvif-media.c:
3169 rtsp-server: Switch around sendonly/recvonly attributes
3170 They are wrong in the ONVIF streaming spec. The backchannel should be
3171 recvonly and the normal media should be sendonly: direction is always
3172 from the point of view of the SDP offerer (the server) according to
3175 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
3177 * docs/libs/gst-rtsp-server-docs.sgml:
3178 * docs/libs/gst-rtsp-server-sections.txt:
3179 * examples/.gitignore:
3180 * examples/Makefile.am:
3181 * examples/test-onvif-backchannel.c:
3182 * gst/rtsp-server/Makefile.am:
3183 * gst/rtsp-server/rtsp-media.h:
3184 * gst/rtsp-server/rtsp-onvif-client.c:
3185 * gst/rtsp-server/rtsp-onvif-client.h:
3186 * gst/rtsp-server/rtsp-onvif-media-factory.c:
3187 * gst/rtsp-server/rtsp-onvif-media-factory.h:
3188 * gst/rtsp-server/rtsp-onvif-media.c:
3189 * gst/rtsp-server/rtsp-onvif-media.h:
3190 * gst/rtsp-server/rtsp-onvif-server.c:
3191 * gst/rtsp-server/rtsp-onvif-server.h:
3192 * gst/rtsp-server/rtsp-sdp.c:
3193 * gst/rtsp-server/rtsp-sdp.h:
3194 rtsp: Add support for ONVIF backchannel
3195 This adds a new RTSP server, client, media-factory and media subclass
3196 for handling the specifics of the backchannel. Ideally this later can be
3197 extended with other ONVIF specific features.
3199 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3201 * gst/rtsp-server/rtsp-media.c:
3202 rtsp-media: Add support for sending+receiving medias
3203 We need to add an appsrc/appsink in that case because otherwise the
3204 media bin will be a sink and a source for rtpbin, causing a pipeline
3206 https://bugzilla.gnome.org/show_bug.cgi?id=788950
3208 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3214 === release 1.13.1 ===
3216 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3220 * gst-rtsp-server.doap:
3224 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3226 * gst/rtsp-server/rtsp-session-pool.c:
3227 session-pool: remove nullable return annotation
3228 create_watch can only return NULL from the API guards, no
3231 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3233 * gst/rtsp-server/rtsp-media-factory.c:
3234 * gst/rtsp-server/rtsp-media.c:
3235 set_clock functions: Add nullable annotations
3237 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3239 * gst/rtsp-server/rtsp-auth.c:
3240 * gst/rtsp-server/rtsp-client.c:
3241 * gst/rtsp-server/rtsp-media-factory.c:
3242 * gst/rtsp-server/rtsp-media.c:
3243 * gst/rtsp-server/rtsp-mount-points.c:
3244 * gst/rtsp-server/rtsp-server.c:
3245 * gst/rtsp-server/rtsp-session-media.c:
3246 * gst/rtsp-server/rtsp-session-pool.c:
3247 * gst/rtsp-server/rtsp-session.c:
3248 * gst/rtsp-server/rtsp-stream-transport.c:
3249 * gst/rtsp-server/rtsp-stream.c:
3250 * gst/rtsp-server/rtsp-thread-pool.c:
3251 All around: add annotations and API guards
3253 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3255 * tests/test-cleanup.c:
3256 test-cleanup: bind any port
3257 The meson test suite runs tests in parallel, trying to bind
3258 a single port made the test fail.
3260 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
3263 meson: make version numbers ints and fix int/string comparison
3264 WARNING: Trying to compare values of different types (str, int).
3265 The result of this is undefined and will become a hard error
3266 in a future Meson release.
3268 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3270 * gst/rtsp-server/rtsp-context.c:
3271 gst_rtsp_context_get_current: add (skip) annotation
3272 The return value type is defined with G_DEFINE_POINTER_TYPE,
3273 and gi emits the following warning:
3274 Invalid non-constant return of bare structure or union; register as
3275 boxed type or (skip)
3277 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
3279 * gst/rtsp-server/rtsp-client.c:
3280 rtsp-client: add type annotations
3281 gi doesn't seem to be able to figure out the type of the
3282 signal parameters when defined with G_DEFINE_POINTER_TYPE
3284 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3287 autotools: use -fno-strict-aliasing where supported
3288 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3290 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3293 meson: use -fno-strict-aliasing where supported
3294 https://bugzilla.gnome.org/show_bug.cgi?id=769183
3296 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3298 * gst/rtsp-server/rtsp-mount-points.c:
3299 mount-points: bail out of loop again when matching mount points
3300 Previous patch led to us iterating the entire sequence. Bail out
3301 of the loop again if we have a match but are moving away from it.
3302 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3304 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3306 * tests/check/gst/mountpoints.c:
3307 tests: mountpoints: add more checks for mount point path matching
3308 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3310 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
3312 * gst/rtsp-server/rtsp-mount-points.c:
3313 mount-points: fix matching of paths where there's also an entry with a common prefix
3314 e.g. with the following mount points
3318 _match() would not match /raw/video and /raw/snapshot correctly.
3319 https://bugzilla.gnome.org/show_bug.cgi?id=771555
3321 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3323 * docs/libs/gst-rtsp-server-sections.txt:
3324 * gst/rtsp-server/rtsp-permissions.c:
3325 * gst/rtsp-server/rtsp-permissions.h:
3326 * tests/check/gst/permissions.c:
3327 permissions: add some new API to make this usable from bindings
3328 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3330 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
3332 * gst/rtsp-server/rtsp-token.c:
3333 rtsp-token: annotate constructors for bindings
3334 This maps _new_empty() to _new(), which also makes RTSPToken()
3335 work properly now. Since this API wasn't usable from bindings
3336 before, this should hopefully be fine.
3337 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3339 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3341 * docs/libs/gst-rtsp-server-sections.txt:
3342 * gst/rtsp-server/rtsp-token.c:
3343 * gst/rtsp-server/rtsp-token.h:
3344 * tests/check/gst/token.c:
3345 rtsp-token: add some API to set fields from bindings
3346 The existing functions are all vararg-based and as such
3347 not usable from bindings.
3348 https://bugzilla.gnome.org/show_bug.cgi?id=787073
3350 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
3352 * tests/check/gst/rtspclientsink.c:
3353 * tests/check/gst/rtspserver.c:
3354 * tests/check/gst/sessionpool.c:
3355 * tests/check/gst/stream.c:
3356 tests: fix indentation
3359 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
3361 * tests/check/gst/rtspserver.c:
3362 tests: rtspserver: fix another ref leak
3363 Even if this didn't show up in valgrind.
3365 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3367 * tests/check/gst/rtspclientsink.c:
3368 tests: rtspclientsink: fix leak
3370 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
3372 * tests/check/gst/rtspserver.c:
3373 test: rtspserver: plug memory leak in test_no_session_timeout
3374 In test_no_session_timeout, unref the rtsp session object when the
3376 https://bugzilla.gnome.org/show_bug.cgi?id=792127
3378 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
3380 * gst/rtsp-sink/gstrtspclientsink.c:
3381 rtpsclientsink: Initialize and clear newly added mutex and cond
3382 While it *did* work, glib would automatically create new mutex and cond
3383 ... which never got freed
3385 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3387 * gst/rtsp-server/rtsp-stream.c:
3388 rtsp-stream: Set multicast TTL on the multicast sockets
3389 And not if we do unicast UDP.
3390 https://bugzilla.gnome.org/show_bug.cgi?id=791743
3392 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
3394 * gst/rtsp-server/rtsp-stream.c:
3395 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
3396 In the multicast case (as in test-multicast, not test-multicast2), the
3397 address could be allocated/reserved (and thus set) already without
3398 allocating the actual socket. We need to allocate the socket here still
3399 instead of just claiming that it was already allocated.
3400 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
3402 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3404 * gst/rtsp-sink/gstrtspclientsink.c:
3405 * gst/rtsp-sink/gstrtspclientsink.h:
3406 rtspclientsink: Use the new rtsp-stream API
3407 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3409 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3411 * gst/rtsp-sink/gstrtspclientsink.c:
3412 * gst/rtsp-sink/gstrtspclientsink.h:
3413 rtspclientsink: Wait until OPEN has been scheduled
3414 Make sure that the sink thread has started opening connection
3415 to the server before continuing.
3416 https://bugzilla.gnome.org/show_bug.cgi?id=790412
3418 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
3421 Automatic update of common submodule
3422 From e8c7a71 to 3fa2c9e
3424 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
3426 * gst/rtsp-server/rtsp-media.c:
3427 * gst/rtsp-server/rtsp-session-media.c:
3428 * gst/rtsp-server/rtsp-stream.c:
3429 rtsp-server: Minor doc fixes
3432 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3435 * tests/Makefile.am:
3436 tests: disable all tests when --disable-tests is used
3437 Move conditional subdir include into top level.
3438 Based on patch by: Joel Holdsworth
3439 https://bugzilla.gnome.org/show_bug.cgi?id=757703
3441 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
3444 * meson_options.txt:
3445 * tests/meson.build:
3446 meson: build more tests and add options to disable tests and examples
3448 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
3450 * gst/rtsp-server/rtsp-session.c:
3451 Fix build when -Werror=deprecated-declarations is on
3452 As gst_rtsp_session_next_timeout is deprecated.
3454 ../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]
3455 res = (gst_rtsp_session_next_timeout (session, now) == 0);
3457 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
3458 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
3459 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
3462 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
3465 Automatic update of common submodule
3466 From 3f4aa96 to e8c7a71
3468 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3470 * tests/check/gst/media.c:
3471 check/media: Add seekability test case: not all streams are active
3472 Media contains two streams but only one is complete and prepared
3474 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3476 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3478 * gst/rtsp-server/rtsp-stream.c:
3479 rtsp-stream: Do not reset 'blocking' if stream is already blocked
3480 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3482 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
3484 * gst/rtsp-server/rtsp-media.c:
3485 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
3486 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3488 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
3491 meson: remove vs_module_defs_dir variable which is no longer needed
3493 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
3495 * gst/rtsp-server/rtsp-session.h:
3498 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
3501 * gst/rtsp-server/meson.build:
3503 * win32/common/libgstrtspserver.def:
3504 win32: remove .def file with exports
3505 They're no longer needed, symbol exporting is now explicit
3506 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
3508 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3511 autotools: stop controlling symbol visibility with -export-symbols-regex
3512 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
3513 This should result in consistent behaviour for the autotools and
3516 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
3518 * gst/rtsp-server/rtsp-media.h:
3519 * gst/rtsp-server/rtsp-server.h:
3520 * gst/rtsp-server/rtsp-session.c:
3521 * gst/rtsp-server/rtsp-session.h:
3522 rtsp-server: add missing GST_EXPORT and export deprecated funcs
3524 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
3526 * tests/check/gst/media.c:
3527 check: Add seekability testing on medias
3528 Make sure that once GstRTSPMedia are prepared they returned
3529 the expected seekability results
3530 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3532 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
3534 * docs/libs/gst-rtsp-server-sections.txt:
3535 * gst/rtsp-server/rtsp-media.c:
3536 * gst/rtsp-server/rtsp-stream.c:
3537 * gst/rtsp-server/rtsp-stream.h:
3538 * win32/common/libgstrtspserver.def:
3539 rtsp-media: Enable seeking query before pipeline is complete
3540 SDP are now provided *before* the pipeline is fully complete. In order
3541 to know whether a media is seekable or not therefore requires asking
3542 the invididual streams.
3543 API: gst_rtsp_stream_seekable
3544 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3546 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
3548 * gst/rtsp-server/rtsp-media.c:
3549 rtsp-media: Fix handling in default_unsuspend()
3550 Handle the case when streams are not blocked and media
3551 is suspended from PAUSED.
3552 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
3553 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3555 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
3557 * tests/check/gst/media.c:
3558 check/media: Fix thread pool leak.
3559 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
3560 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3562 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
3564 * gst/rtsp-server/rtsp-media.c:
3565 rtsp-media: Removed fakesink elements
3566 There is not need of adding fakesink elements to the media
3567 pipeline in the dynamic-payloader case.
3568 The media pipeline itself is dynamically updated with
3569 the receiver and sender parts that are based on the client
3570 transport information known after SETUP has been received.
3571 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
3572 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3574 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
3576 * gst/rtsp-server/rtsp-media.c:
3577 rtsp-media: Corrected ASYNC_DONE handling
3578 Media is complete when all the transport based parts are
3579 added to the media pipeline. At this point ASYNC_DONE is
3580 posted by the media pipeline and media is ready to enter
3582 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
3583 https://bugzilla.gnome.org/show_bug.cgi?id=790674
3585 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
3587 * tests/check/gst/media.c:
3588 check/media: Check that prepared media can provide a SDP
3589 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
3591 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
3593 * gst/rtsp-server/rtsp-client.c:
3594 rtsp-client: Don't leak addr
3597 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
3599 * gst/rtsp-server/rtsp-client.c:
3600 * gst/rtsp-server/rtsp-session-media.c:
3601 * gst/rtsp-server/rtsp-stream.c:
3604 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
3606 * gst/rtsp-server/rtsp-media.c:
3607 rtsp-media: Don't unblock with remaining dynamic payloaders
3608 If we still have some dynamic paylaoders which haven't posted
3609 no-more-pads yet, don't go to PREPARED if one of the streams
3611 The risk was that we would end up not exposing/using all specified
3613 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
3614 then it will take a bit more time to start. But only if those 3
3615 conditions are present.
3616 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3618 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
3620 * gst/rtsp-server/rtsp-media.c:
3623 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
3625 * gst/rtsp-server/rtsp-media.c:
3626 rtsp-media: Don't set float on a gint64 variable
3627 Just use 0. Fixes 'undefined' behaviour from clang
3629 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
3631 * gst/rtsp-server/rtsp-media.c:
3632 rtsp-media: Fix previous commit
3633 We only want to count dynamic payloaders
3635 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
3637 * gst/rtsp-server/rtsp-media.c:
3638 * tests/check/gst/media.c:
3639 rtsp-media: Handle multiple dynamic elements
3640 If we have more than one dynamic payloader in the pipeline, we need
3641 to wait until the *last* one emits 'no-more-pads' before switching
3643 Failure to do so would result in a race where some of the streams
3644 wouldn't properly be prepared
3645 https://bugzilla.gnome.org/show_bug.cgi?id=769521
3647 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
3649 * win32/common/libgstrtspserver.def:
3650 win32: Fix exported symbols list
3652 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3654 * gst/rtsp-server/rtsp-stream.c:
3655 rtsp-stream: Only update the RTP udpsink if it actually exists
3656 For send-only streams it does not exist, but the RTCP udpsink might.
3658 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
3660 * win32/common/libgstrtspserver.def:
3661 win32: Update exports
3663 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
3665 * gst/rtsp-server/rtsp-media.c:
3666 * gst/rtsp-server/rtsp-stream.c:
3667 * gst/rtsp-server/rtsp-stream.h:
3668 rtsp-media: seek on media pipelines that are complete
3669 Make sure that a seek is performed on pipelines that
3670 contain at least one sink element.
3671 Change-Id: Icf398e10add3191d104b1289de612412da326819
3672 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3674 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
3676 * gst/rtsp-server/rtsp-client.c:
3677 * gst/rtsp-server/rtsp-media.c:
3678 * gst/rtsp-server/rtsp-media.h:
3679 * gst/rtsp-server/rtsp-stream.c:
3680 * gst/rtsp-server/rtsp-stream.h:
3681 * tests/check/gst/client.c:
3682 * tests/check/gst/media.c:
3683 * tests/check/gst/rtspserver.c:
3684 * tests/check/gst/stream.c:
3685 Dynamically reconfigure pipeline in PLAY based on transports
3686 The initial pipeline does not contain specific transport
3687 elements. The receiver and the sender parts are added
3689 If the media is shared, the streams are dynamically
3690 reconfigured after each PLAY.
3691 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3693 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
3695 * gst/rtsp-server/rtsp-stream.c:
3696 rtsp-stream: obtain stream position from pad
3697 If no sinks have been added yet, obtain the current and
3698 the stop position of the stream from the send_src pad.
3699 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
3700 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3702 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
3704 * gst/rtsp-server/rtsp-session-media.c:
3705 * gst/rtsp-server/rtsp-session-media.h:
3706 rtsp-session-media: add function to get a list of transports
3707 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
3708 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3710 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
3712 * gst/rtsp-server/rtsp-stream.c:
3713 * gst/rtsp-server/rtsp-stream.h:
3714 rtsp-stream: add functions to get rtp and rtcp multicast sockets
3715 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
3716 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3718 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
3720 * gst/rtsp-server/rtsp-stream.c:
3721 stream: set async=sync=false only for RTCP appsink
3722 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
3723 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3725 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
3727 * gst/rtsp-server/rtsp-media.c:
3728 rtsp-media: return minimum value in query position case
3729 The minimum position should be returned as we are interested
3730 in the whole interval.
3731 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
3732 https://bugzilla.gnome.org/show_bug.cgi?id=788340
3734 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
3736 * gst/rtsp-server/rtsp-session.c:
3737 * tests/check/gst/rtspserver.c:
3738 rtsp-session: Handle the case when timeout=0
3739 According to the documentation, a timeout of value 0 means
3740 that the session never timeouts. This adds handling of that.
3741 If timeout=0 we just return with a -1 from
3742 gst_rtsp_session_next_timeout_usec ().
3743 https://bugzilla.gnome.org/show_bug.cgi?id=785058
3745 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3747 * gst/rtsp-sink/gstrtspclientsink.c:
3748 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
3749 https://bugzilla.gnome.org/show_bug.cgi?id=785024
3751 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
3753 * docs/libs/gst-rtsp-server-sections.txt:
3754 * gst/rtsp-server/rtsp-media-factory.c:
3755 docs: add media factory transport mode accessors
3756 and fix the documentation for the return value of the getter
3758 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
3760 * gst/rtsp-server/rtsp-client.c:
3761 rtsp-client: unref 'pipelined_requests' in finalize
3762 The hash table priv->pipelined_requests is not unref:ed in the
3763 finalize funktion. Make sure it is.
3764 https://bugzilla.gnome.org/show_bug.cgi?id=788704
3766 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
3768 * gst/rtsp-server/rtsp-media.c:
3769 rtsp-media: Initialize scalar variable
3772 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
3774 * win32/common/libgstrtspserver.def:
3775 win32: Update export file
3777 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
3779 * gst/rtsp-server/rtsp-client.c:
3780 * gst/rtsp-server/rtsp-media.c:
3781 * gst/rtsp-server/rtsp-media.h:
3782 Start support for RTSP 2.0
3783 This adds basic support for new 2.0 features, though the protocol is
3784 subposdely backward incompatible, most semantics are the sames.
3787 * version negotiation
3788 * pipelined requests support
3789 * Media-Properties support
3790 * Accept-Ranges support
3792 * gst_rtsp_media_seekable
3793 The RTSP methods that have been removed when using 2.0 now return
3795 https://bugzilla.gnome.org/show_bug.cgi?id=781446
3797 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3799 * gst/rtsp-server/rtsp-stream.c:
3800 stream: Use stream duration as stream-stop if segment was not configured with a stop
3801 Allowing client to know stream duration when no seeking happened.
3802 https://bugzilla.gnome.org/show_bug.cgi?id=783435
3804 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
3806 * gst/rtsp-server/rtsp-media-factory.c:
3807 rtsp-media-factory: Don't cache any media if NULL was returned as key
3808 The docs already mentioned this, but we actually stored it in the hash
3809 table with key==NULL and leaked its reference forever.
3811 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
3813 * gst/rtsp-sink/gstrtspclientsink.c:
3814 * gst/rtsp-sink/gstrtspclientsink.h:
3815 rtspclientsink: Use a mutex for protecting against concurrent send/receives
3816 This is a simple port of:
3817 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
3818 * c438545dc9e2f14f657bc0ef261fff726449867b
3819 * cd17c71dcea5c9310d21f1347c7520983e5869ac
3820 in gst-plugins-good.
3822 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
3824 * gst/rtsp-server/rtsp-sdp.c:
3825 sdp: fix Memory leak in error case
3826 https://bugzilla.gnome.org/show_bug.cgi?id=787059
3828 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
3830 * pkgconfig/meson.build:
3831 meson: don't install -uninstalled.pc file
3832 https://bugzilla.gnome.org/show_bug.cgi?id=786457
3834 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
3837 Automatic update of common submodule
3838 From 48a5d85 to 3f4aa96
3840 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3842 * gst/rtsp-server/rtsp-client.c:
3843 rtsp-client: Fix typo in debug message
3845 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
3848 meson: hide symbols by default unless explicitly exported
3850 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
3852 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3853 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
3854 Fixes meson warning about undefined @srcdir@.
3856 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
3858 * tests/meson.build:
3859 meson: skip tests on windows for now
3860 As we do in the other modules. As libgstcheck is currently not
3861 built on windows. Fixes "Fallback variable 'gst_check_dep' in
3862 the subproject 'gstreamer' does not exist"" Meson error.
3864 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
3866 * gst/rtsp-server/rtsp-stream.c:
3867 rtsp-stream: fix connection delay due to wrong assumption on last-sample
3868 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
3869 multiudpsink's last-sample always comes from the payloader. Which
3870 is wrong if auxiliary streams are multiplexed in the same stream.
3871 So check the buffer's ssrc against the caps'ssrc before to use its
3872 seqnum. If not the same ssrc just use the payloader as done prior
3873 the commit above or when there is no last-sample yet.
3874 https://bugzilla.gnome.org/show_bug.cgi?id=784094
3876 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3879 meson: Allow using glib as a subproject
3881 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
3884 meson: fix with-package-name option
3885 https://bugzilla.gnome.org/show_bug.cgi?id=784082
3887 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3890 Distribute meson_options.txt
3892 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3895 And config.h.meson is no longer dist either
3897 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
3901 meson: config.h.meson is no longer needed
3903 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3905 * tests/check/meson.build:
3906 * tests/meson.build:
3907 meson: Fix building tests and activate them again
3909 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
3911 * tests/check/meson.build:
3912 meson: Do not use path separator in test names
3913 Avoiding warnings like:
3914 WARNING: Target "elements/audioamplify" has a path separator in its name.
3916 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
3919 * meson_options.txt:
3920 meson: add options to set package name and origin
3921 https://bugzilla.gnome.org/show_bug.cgi?id=782172
3923 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
3925 * gst/rtsp-server/rtsp-address-pool.h:
3926 * gst/rtsp-server/rtsp-auth.h:
3927 * gst/rtsp-server/rtsp-client.h:
3928 * gst/rtsp-server/rtsp-context.h:
3929 * gst/rtsp-server/rtsp-media-factory-uri.h:
3930 * gst/rtsp-server/rtsp-media-factory.h:
3931 * gst/rtsp-server/rtsp-media.h:
3932 * gst/rtsp-server/rtsp-mount-points.h:
3933 * gst/rtsp-server/rtsp-params.h:
3934 * gst/rtsp-server/rtsp-permissions.h:
3935 * gst/rtsp-server/rtsp-sdp.h:
3936 * gst/rtsp-server/rtsp-server.h:
3937 * gst/rtsp-server/rtsp-session-media.h:
3938 * gst/rtsp-server/rtsp-session-pool.h:
3939 * gst/rtsp-server/rtsp-session.h:
3940 * gst/rtsp-server/rtsp-stream-transport.h:
3941 * gst/rtsp-server/rtsp-stream.h:
3942 * gst/rtsp-server/rtsp-thread-pool.h:
3943 * gst/rtsp-server/rtsp-token.h:
3944 Mark symbols explicitly for export with GST_EXPORT
3946 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3949 * gst/rtsp-sink/Makefile.am:
3950 Remove plugin specific static build option
3951 Static and dynamic plugins now have the same interface. The standard
3952 --enable-static/--enable-shared toggle are sufficient.
3954 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
3960 === release 1.12.0 ===
3962 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
3968 * gst-rtsp-server.doap:
3972 === release 1.11.91 ===
3974 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
3980 * gst-rtsp-server.doap:
3984 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
3987 Automatic update of common submodule
3988 From 60aeef6 to 48a5d85
3990 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
3992 * gst/rtsp-server/rtsp-media-factory.c:
3993 * gst/rtsp-server/rtsp-media.c:
3994 * gst/rtsp-server/rtsp-session.c:
3995 * gst/rtsp-server/rtsp-stream.c:
3996 gi: Fix some annotations and docstrings
3998 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4000 * gst/rtsp-server/meson.build:
4002 * meson_options.txt:
4005 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
4009 Automatic update of common submodule
4010 From 39ac2f5 to 60aeef6
4012 === release 1.11.90 ===
4014 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
4020 * gst-rtsp-server.doap:
4024 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
4026 * examples/test-launch.c:
4027 examples: make test-launch pipeline shared by default as well
4029 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
4031 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4032 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
4033 Just the build dir is not going to work for srcdir!=builddir.
4035 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
4038 meson: Update version
4040 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4045 === release 1.11.2 ===
4047 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4053 * gst-rtsp-server.doap:
4056 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
4059 meson: dist meson build files
4060 Ship meson build files in tarballs, so people who use tarballs
4061 in their builds can start playing with meson already.
4063 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
4065 * examples/test-record.c:
4066 examples/test-record: Add extra line to initial printout
4067 Add an example line of how to deliver a stream to the
4070 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4072 * gst/rtsp-server/rtsp-client.c:
4073 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
4074 If there is no Content-Length header, no body would be allocated and the
4075 '\0' would also not be appended to the body.
4077 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
4079 * gst/rtsp-server/rtsp-client.c:
4080 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
4081 While they logically have 0 bytes length, GstRTSPConnection is appending
4082 a '\0' to everything making the size be 1 instead.
4084 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4089 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
4091 * gst/rtsp-server/rtsp-session.c:
4092 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
4093 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
4096 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
4101 === release 1.11.1 ===
4103 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4109 * gst-rtsp-server.doap:
4110 * win32/common/libgstrtspserver.def:
4113 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
4115 * gst/rtsp-server/rtsp-stream.c:
4116 rtsp-stream: corrected if-statement in _get_server_port()
4117 This bug was accidentally introduced while fixing a segfault
4118 in _get_server_port() function.
4119 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4121 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
4123 * gst/rtsp-server/rtsp-stream.c:
4124 * tests/check/gst/stream.c:
4125 rtsp-stream: fixed segmenation fault in _get_server_port()
4126 Calling function gst_rtsp_stream_get_server_port() results in
4127 segmenation fault in the RTP/RTSP/TCP case.
4128 Port that the server will use to receive RTCP makes only
4129 sense in the UDP case, however the function should handle
4130 the TCP case in a nicer way.
4131 https://bugzilla.gnome.org/show_bug.cgi?id=776345
4133 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
4135 * gst/rtsp-server/rtsp-media-factory.c:
4136 dosc: Fix a little typo
4137 https://bugzilla.gnome.org/show_bug.cgi?id=777037
4139 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4141 * pkgconfig/Makefile.am:
4142 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4143 * pkgconfig/meson.build:
4144 meson: generate pkg-config -uninstalled pc files
4145 Generating those files is useful for users building the GStreamer stack
4146 using meson and having to link it to another project which is still
4147 using the autotools.
4148 https://bugzilla.gnome.org/show_bug.cgi?id=776810
4150 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
4152 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4153 pkgconfig: fix -uninstalled pc file
4154 pcfiledir was never defined so the paths were wrong.
4155 https://bugzilla.gnome.org/show_bug.cgi?id=776867
4157 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
4159 * gst/rtsp-server/rtsp-stream.c:
4160 * tests/check/gst/rtspserver.c:
4161 rtsp-stream: Fixed TCP transport case
4162 Make sure that the appsink element is actually added to
4163 the bin before trying to link it with the elements in it.
4164 https://bugzilla.gnome.org/show_bug.cgi?id=776343
4166 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4172 Remove generated .spec file
4173 Likely extremely bitrotten, and we should not ship this anyway.
4175 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
4178 Automatic update of common submodule
4179 From f980fd9 to 39ac2f5
4181 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
4183 * gst/rtsp-server/rtsp-media.c:
4184 media: Fix pt map caps
4185 Since decryption is handled within rtpbin, all outcoming stream
4186 caps will be application/x-rtp (i.e. regular rtp)
4187 Fixes RECORD with SRTP streams
4189 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
4191 * gst/rtsp-server/rtsp-media-factory.c:
4192 media-factory: Create media objects with the proper transport mode
4193 The function called immediately afterwards (collect_streams()) will
4194 need it to work properly
4196 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
4198 * gst/rtsp-server/rtsp-auth.c:
4199 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
4201 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
4203 * gst/rtsp-server/rtsp-media-factory.c:
4204 rtsp-media-factory: Don't create a pipeline for the media pipeline string
4205 We're going to put a pipeline into a pipeline otherwise, which is not
4208 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
4210 * gst/rtsp-server/rtsp-media.c:
4211 media: Fix race condition around finish_unprepare() if called multiple time
4212 https://bugzilla.gnome.org/show_bug.cgi?id=755329
4214 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
4216 * gst/rtsp-sink/gstrtspclientsink.c:
4217 rtspclientsink: Don't leave stale pointer after unref
4218 Fix a warning on shutdown - don't keep a pointer to an
4219 alread-unreffed object.
4221 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4224 common: use https protocol for common submodule
4225 https://bugzilla.gnome.org/show_bug.cgi?id=775110
4227 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
4229 * gst/rtsp-server/rtsp-stream.c:
4230 stream: block the output of rtpbin instead of the source pipeline
4231 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
4232 detection of the srtp rollover counter to add to the SDP.
4233 Unfortunately, it was incomplete for live pipelines where the logic
4234 blocks the source bin before creating the SDP and thus would never have
4235 the necessary informaiton to create a correct SDP with srtp encryption.
4236 Move the pad blocks to rtpbin's output pads instead so that the
4237 necessary information can be created before we need the information for
4239 https://bugzilla.gnome.org/show_bug.cgi?id=770239
4241 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
4243 * gst/rtsp-server/rtsp-client.c:
4244 rtsp-client: add IDLE timeout, before session exists
4245 The RTSP server will not timeout an idle RTSP connection
4246 (note this is different from doing timeout on a RTSP
4248 At least for Apache this is a problem when running RTSP over
4249 HTTPS since it uses one of the threads (there is a rather
4250 limited number) that are available for handling requests.
4251 https://bugzilla.gnome.org/show_bug.cgi?id=771830
4253 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
4258 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
4260 * gst/rtsp-server/rtsp-stream.c:
4261 rtsp-stream: Set close-socket FALSE on UDP src:es
4262 With this RTSP server can use the sockets independent on the udpsrc
4264 When the udp src is finalized it will unref socket and when g_socket
4265 is finalized the socket will be closed.
4266 https://bugzilla.gnome.org/show_bug.cgi?id=765673
4268 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4270 * gst/rtsp-sink/gstrtspclientsink.c:
4271 rtspclientsink: Move to new helper function to parse authentication responses
4272 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4274 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4276 * examples/Makefile.am:
4277 * examples/test-auth-digest.c:
4278 * gst/rtsp-server/rtsp-auth.c:
4279 * gst/rtsp-server/rtsp-auth.h:
4280 * win32/common/libgstrtspserver.def:
4281 rtsp-auth: Add support for Digest authentication
4282 https://bugzilla.gnome.org/show_bug.cgi?id=774416
4284 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4287 * gst/rtsp-server/meson.build:
4289 * tests/check/meson.build:
4291 * win32/common/libgstrtspserver.def:
4292 Enable building with MSVC
4293 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4295 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
4298 meson: gstreamer gst_check_dep does not exist on windows
4300 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
4302 * gst/rtsp-server/rtsp-client.c:
4303 client: update do_send_message to match type GstRTSPClientSendFunc
4304 This type mismatch fails building with MSVC
4305 https://bugzilla.gnome.org/show_bug.cgi?id=774640
4307 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4309 * gst/rtsp-server/rtsp-sdp.c:
4310 rtsp-sdp: Fix indentation
4312 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
4314 * gst/rtsp-server/rtsp-media.c:
4315 rtsp-media: Only signal "new-state" if the state has actually changed
4316 https://bugzilla.gnome.org/show_bug.cgi?id=774173
4318 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
4320 * gst/rtsp-server/rtsp-client.c:
4321 * gst/rtsp-server/rtsp-client.h:
4322 client: emit signal in the beginning of each rtsp request
4323 These signals let the application validate the requests, configure the
4324 media/stream in a certain way and also generate error status code in
4325 case of error or bad request.
4326 https://bugzilla.gnome.org/show_bug.cgi?id=758062
4328 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
4331 meson: update version
4333 === release 1.11.0 ===
4335 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4340 === release 1.10.0 ===
4342 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4348 * gst-rtsp-server.doap:
4351 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
4353 * tests/check/gst/rtspserver.c:
4354 * tests/check/gst/stream.c:
4355 tests: try to avoid using the same ports in different tests
4356 Causes problems with client multicast tests otherwise if
4357 tests are run in parallel.
4358 https://bugzilla.gnome.org/show_bug.cgi?id=773640
4360 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4362 * tests/check/gst/client.c:
4363 tests: client: use fail_unless_equals_foo() for better failure reporting
4365 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
4367 * gst/rtsp-server/rtsp-client.c:
4368 rtsp-client: Session filter in unwatch session
4369 Call session filter with filter_session_media as paramer in
4370 client_unwatch_session if using drop_backlog = FALSE.
4371 In client_unwatch_session its allowed to grow the watchs backlog.
4372 If using drop_backlog = FALSE and the backlog is full it will cause
4373 a deadlock when setting session media state to NULL
4374 if the backlog is not allowed to grow.
4375 https://bugzilla.gnome.org/show_bug.cgi?id=771983
4377 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
4380 meson: add fallbacks for gst modules
4383 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
4385 * gst/rtsp-server/rtsp-client.c:
4386 rtsp-client: Fix factory leaking in find_media() in error cases
4387 https://bugzilla.gnome.org/show_bug.cgi?id=771488
4389 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4391 * gst/rtsp-server/rtsp-stream.c:
4392 stream: Fix randomly missing streams from SDP with dynamic elements
4393 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
4394 "pad-added" signal. In that case priv->srcpad could already have its caps,
4395 and they'll be sent to priv->send_src[0] pad. That means that when it
4396 connects "notify::caps" signal, that pad could already have received its
4397 caps and the signal won't be emitted anymore.
4398 In that case priv->caps stay to NULL and when building the SDP that stream
4399 gets ignored. Leading to missing video or audio when playing in client side.
4400 https://bugzilla.gnome.org/show_bug.cgi?id=772478
4402 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
4405 meson: update version
4407 === release 1.9.90 ===
4409 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4415 * gst-rtsp-server.doap:
4418 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
4420 * gst/rtsp-server/rtsp-media-factory.c:
4421 * gst/rtsp-server/rtsp-media.c:
4422 * gst/rtsp-server/rtsp-stream.c:
4423 rtsp-server: Hint that set_multicast_iface expects the name of the interface
4424 To prevent any possibly confusion with IPs or anything else.
4425 https://bugzilla.gnome.org/show_bug.cgi?id=771530
4427 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
4429 * gst/rtsp-server/rtsp-media-factory.c:
4430 * gst/rtsp-server/rtsp-media.c:
4431 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
4432 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
4434 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
4437 configure: Depend on gstreamer 1.9.2.1
4439 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
4443 Automatic update of common submodule
4444 From b18d820 to f980fd9
4446 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
4450 Automatic update of common submodule
4451 From 6f2d209 to b18d820
4453 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
4455 * gst/rtsp-server/rtsp-stream.c:
4456 rtsp-stream: Remove unused _locked() variant of a function
4457 It was added during refactoring.
4459 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4461 * gst/rtsp-server/rtsp-stream.c:
4462 stream: cosmetic cleanup
4463 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4465 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4467 * gst/rtsp-server/rtsp-stream.c:
4468 stream: Compare IP addresses case insensitive in more places
4469 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4471 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4474 * gst/rtsp-server/rtsp-stream.c:
4475 stream: Fix leaked joined_bin
4476 There is no need to keep a strong ref on it, and _leave_bin() was
4477 setting it to NULL before calling g_clear_object() so it was leaked.
4478 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4480 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4482 * gst/rtsp-server/rtsp-stream.c:
4483 rtsp-stream: Compare IP address strings case insensitive
4484 Otherwise IPv6 addresses might fail this comparision.
4486 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
4488 * gst/rtsp-server/rtsp-stream.c:
4489 rtsp-stream: Bind multicast sockets to ANY as before
4490 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
4492 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
4494 * gst/rtsp-server/rtsp-session.c:
4495 rtsp-session: Fix segfault when doing keep-alive after removing the session
4496 If keep-alive happens after removing the session but before finalizing the
4497 stream transport, we would segfault.
4498 https://bugzilla.gnome.org/show_bug.cgi?id=750544
4500 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
4502 * gst/rtsp-server/rtsp-stream.c:
4503 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
4504 Adding them later will cause deadlocks due to
4505 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
4506 2) adding the multicast sink
4507 3) waiting for it to get data to preroll again
4508 3) never happens because the queues after the tee are full.
4510 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
4512 * gst/rtsp-server/rtsp-stream.c:
4513 rtsp-stream: Fix up various multicast related issues
4515 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
4517 * tests/check/gst/stream.c:
4518 tests: Fix compilation
4520 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4522 * gst/rtsp-server/rtsp-client.c:
4523 * gst/rtsp-server/rtsp-stream.c:
4524 * tests/check/gst/stream.c:
4525 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
4526 This is basically reverting changes introduced in commit f62a9a7,
4527 because it was introducing various regressions:
4528 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
4529 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
4530 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
4531 - If a mcast client connects, it creates a new socket in SETUP to try to respect
4532 the destination/port given by the client in the transport, and overrides the
4533 socket already set on the udpsink element. That means that if we already had a
4534 client connected, the source address on the udp packets it receives suddenly
4536 - If a 2nd mcast client connects, the destination/port in its transport is
4537 ignored but its transport wasn't updated.
4538 What this patch does:
4539 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
4540 - Always have a tee+queue when udp is enabled. This could be optimized
4541 again in a later patch, but is more complicated. If no unicast clients
4542 connects then those elements are useless, this could be also optimized
4544 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
4545 seperated from those for unicast clients. Since we already support only
4546 one mcast address, we also create only one set of elements.
4547 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4549 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4551 * gst/rtsp-server/rtsp-stream.c:
4552 stream: factor our plug_src function
4553 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4555 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4557 * gst/rtsp-server/rtsp-stream.c:
4558 stream: factor out plug_sink function
4559 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4561 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4563 * gst/rtsp-server/rtsp-stream.c:
4564 stream: small documentation clarification
4565 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4567 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4569 * gst/rtsp-server/rtsp-stream.c:
4570 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
4571 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4573 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4575 * gst/rtsp-server/rtsp-stream.c:
4576 stream: Keep a ref on joined bin
4577 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4579 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4581 * gst/rtsp-server/rtsp-stream.c:
4582 stream: code cleanup
4583 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4585 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4587 * gst/rtsp-server/rtsp-stream.c:
4588 stream: small fix in error code path
4589 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4591 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4593 * gst/rtsp-server/rtsp-stream.c:
4594 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
4595 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
4596 but keeps unit tests.
4597 https://bugzilla.gnome.org/show_bug.cgi?id=766612
4599 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
4604 === release 1.9.2 ===
4606 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
4612 * gst-rtsp-server.doap:
4615 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
4618 * examples/meson.build:
4620 * gst/rtsp-server/meson.build:
4621 * gst/rtsp-sink/meson.build:
4623 * pkgconfig/meson.build:
4624 * tests/check/meson.build:
4625 * tests/meson.build:
4626 Add support for Meson as alternative/parallel build system
4627 https://github.com/mesonbuild/meson
4629 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
4632 * tests/check/Makefile.am:
4633 build: silence error about pthread for 'make check' in osx
4634 Fixes "clang: error: argument unused during compilation: '-pthread'"
4636 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
4638 * gst/rtsp-server/rtsp-client.c:
4639 rtsp-client: Fix leaking of media in error cases
4640 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
4641 and myself to make the media refcounting a bit easier to follow.
4642 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4644 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
4646 * gst/rtsp-server/rtsp-client.c:
4647 rtsp-client: Fix leaking of session in error cases
4648 https://bugzilla.gnome.org/show_bug.cgi?id=755632
4650 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
4653 Automatic update of common submodule
4654 From f363b32 to f49c55e
4656 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
4661 === release 1.9.1 ===
4663 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
4669 * gst-rtsp-server.doap:
4672 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4675 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
4676 https://bugzilla.gnome.org/show_bug.cgi?id=767463
4678 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4681 Automatic update of common submodule
4682 From ac2f647 to f363b32
4684 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4686 * gst/rtsp-server/rtsp-sdp.c:
4687 * gst/rtsp-server/rtsp-sdp.h:
4688 * gst/rtsp-server/rtsp-stream.c:
4689 * gst/rtsp-server/rtsp-stream.h:
4690 sdp: add rollover counters for all sender SSRC
4691 We add different crypto sessions in MIKEY, one for each sender
4692 SSRC. Currently, all of them will have the same security policy, 0.
4693 The rollover counters are obtained from the srtpenc element using the
4695 https://bugzilla.gnome.org/show_bug.cgi?id=730539
4697 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
4699 * gst/rtsp-server/rtsp-media-factory.h:
4700 * gst/rtsp-server/rtsp-server.h:
4701 docs: fix some typos
4703 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
4705 * gst/rtsp-server/Makefile.am:
4706 g-i: pass compiler env to g-ir-scanner
4707 It's what introspection.mak does as well. Should
4708 fix spurious build failures on gnome-continuous
4709 (caused by g-ir-scanner getting compiler details
4710 via python which is broken in some environments
4711 so passing the compiler details bypasses that).
4713 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
4715 * gst/rtsp-server/rtsp-session.c:
4716 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
4717 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
4718 https://bugzilla.gnome.org/show_bug.cgi?id=766619
4720 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
4722 * gst/rtsp-sink/gstrtspclientsink.c:
4723 rtspclientsink: Check return value of sscanf
4724 And just make sure we always have 0/0 if we have an error
4727 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
4729 * gst/rtsp-server/rtsp-stream.c:
4730 * tests/check/gst/rtspserver.c:
4731 * tests/check/gst/stream.c:
4732 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
4733 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
4734 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
4735 - Create unit test for shared media.
4736 https://bugzilla.gnome.org/show_bug.cgi?id=764744
4738 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
4740 * gst/rtsp-server/rtsp-stream.c:
4741 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
4742 For IPv6 addresses, binding to a multicast group does not work on Linux
4743 either. Always bind to ANY and then later join the multicast group.
4744 https://bugzilla.gnome.org/show_bug.cgi?id=764679
4746 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
4749 Automatic update of common submodule
4750 From 6f2d209 to ac2f647
4752 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
4754 * gst/rtsp-server/rtsp-thread-pool.c:
4755 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
4756 Clarified why it is necessary to add source information to
4757 GstRTSPThreadImpl. See the reported bug in GLib:
4758 https://bugzilla.gnome.org/show_bug.cgi?id=720186
4759 for more information.
4760 https://bugzilla.gnome.org/show_bug.cgi?id=761702
4762 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
4764 * examples/Makefile.am:
4765 examples: Clean up CFLAGS/LDADD even more
4766 The internal .la should come first and is part of LDADD, as is
4769 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
4771 * examples/Makefile.am:
4772 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
4774 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
4776 * gst/rtsp-server/Makefile.am:
4777 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
4779 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
4781 * gst/rtsp-server/rtsp-client.c:
4782 * gst/rtsp-server/rtsp-media-factory.c:
4783 * gst/rtsp-server/rtsp-media-factory.h:
4784 * gst/rtsp-server/rtsp-media.c:
4785 * gst/rtsp-server/rtsp-media.h:
4786 * gst/rtsp-server/rtsp-sdp.c:
4787 * gst/rtsp-server/rtsp-stream.c:
4788 * gst/rtsp-server/rtsp-stream.h:
4789 rtsp-server: Implement clock signalling according to RFC7273
4790 For NTP and PTP clocks we signal the actual clock that is used and signal
4791 the direct media clock offset.
4792 For all other clocks we at least signal that it's the local sender clock.
4793 This allows receivers to know which clock was used to generate the media and
4794 its RTP timestamps. Receivers can then implement network synchronization,
4795 either absolute or at least relative by getting the sender clock rate directly
4796 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
4798 https://bugzilla.gnome.org/show_bug.cgi?id=760005
4800 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
4802 * gst/rtsp-sink/gstrtspclientsink.c:
4803 rtspclientsink: Add support for setting the multicast interface
4804 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4806 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4808 * gst/rtsp-server/rtsp-media-factory.c:
4809 * gst/rtsp-server/rtsp-media-factory.h:
4810 * gst/rtsp-server/rtsp-media.c:
4811 * gst/rtsp-server/rtsp-media.h:
4812 * gst/rtsp-server/rtsp-stream.c:
4813 * gst/rtsp-server/rtsp-stream.h:
4814 rtsp-media: Add support for setting the multicast interface
4815 https://bugzilla.gnome.org/show_bug.cgi?id=763000
4817 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
4819 * gst/rtsp-sink/gstrtspclientsink.c:
4820 rtspclientsink: use new gst_element_class_add_static_pad_template()
4821 https://bugzilla.gnome.org/show_bug.cgi?id=763196
4823 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4828 === release 1.8.0 ===
4830 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
4836 * gst-rtsp-server.doap:
4839 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
4841 * gst/rtsp-server/rtsp-stream.c:
4842 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
4843 This would get us NO_PREROLL in the bin again and break seeking.
4844 Thanks to Carlos Rafael Giani for helping to debug this!
4845 https://bugzilla.gnome.org/show_bug.cgi?id=740509
4847 === release 1.7.91 ===
4849 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
4855 * gst-rtsp-server.doap:
4858 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
4860 * gst/rtsp-server/rtsp-stream.c:
4861 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
4862 Without this, RECORD pipelines are broken because
4863 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
4864 added later. Previously it was there earlier and due to NO_PREROLL caused the
4865 pipeline to preroll immediately
4866 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
4867 as the corresponding code previously was only for PLAY pipelines.
4868 https://bugzilla.gnome.org/show_bug.cgi?id=763281
4870 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
4872 * gst/rtsp-server/rtsp-stream.c:
4873 rtsp-stream: Fix typo in the docstring
4874 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
4876 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
4878 * gst/rtsp-server/rtsp-stream.c:
4879 rtsp-stream: Disable multicast loopback for all our sockets
4880 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
4881 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
4882 loopback setting on the socket... while udpsink does which unfortunately has
4883 no effect here on Windows but on Linux.
4884 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4886 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
4888 * tests/check/gst/stream.c:
4889 stream tests: added new tests
4890 Test a case when the address pool only contains multicast addresses
4891 and the client is requesting unicast udp.
4892 Added tests for multicast ports allocation.
4893 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4895 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
4897 * gst/rtsp-server/rtsp-stream.c:
4898 rtsp-stream: Only bind multicast sockets to ANY on Windows
4899 On Linux it is still needed to bind to the multicast address
4900 to filter out random other packets, while on Windows binding
4901 to multicast addresses just fails.
4903 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
4905 * gst/rtsp-server/rtsp-stream.c:
4906 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
4907 Otherwise we fail to allocate UDP ports if the pool only contains multicast
4908 addresses, which is something that used to work before. For unicast addresses
4909 if the pool contains none, we just allocate them as if there is no pool at
4911 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4913 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
4915 * gst/rtsp-server/rtsp-client.c:
4916 * gst/rtsp-server/rtsp-stream.c:
4917 rtsp-server: Fix indentation
4919 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
4921 * gst/rtsp-server/rtsp-stream.c:
4922 rtsp-stream: Don't bind the sockets to multicast addresses
4923 This works on Linux but fails completely on Windows. You're supposed
4924 to bind to ANY and then join the multicast group.
4925 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4927 === release 1.7.90 ===
4929 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
4935 * gst-rtsp-server.doap:
4938 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
4941 Automatic update of common submodule
4942 From b64f03f to 6f2d209
4944 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
4946 * gst/rtsp-sink/gstrtspclientsink.c:
4947 * tests/check/gst/rtspclientsink.c:
4948 rtspsink: Fix some leaks in rtspclientsink and the unit test.
4949 https://bugzilla.gnome.org/show_bug.cgi?id=762525
4951 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
4953 * tests/check/gst/media.c:
4954 * tests/check/gst/rtspclientsink.c:
4955 * tests/check/gst/rtspserver.c:
4956 * tests/check/gst/stream.c:
4957 tests: unit test fixes
4958 Removed port allocation test from the media suite.
4959 The port allocation failure is now in the stream suite.
4961 Make sure that the media is suspended after the DESCRIBE request
4962 before reconfiguring the UDP sinks.
4964 In the RECORD case we have to set async property to false
4965 for the appsink element in the test in order to make sure
4966 that the media pipeline doesn't hang in start_preroll().
4967 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4969 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
4971 * gst/rtsp-server/rtsp-client.c:
4972 * gst/rtsp-server/rtsp-stream.c:
4973 * gst/rtsp-server/rtsp-stream.h:
4974 rtsp-stream: postpone UDP socket allocation until SETUP
4975 Postpone the allocation of the UDP sockets until we know
4976 what transport has been chosen by the client.
4977 Both unicast and multicast UDP sources are created in one
4979 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4981 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
4983 * gst/rtsp-server/rtsp-stream.c:
4984 rtsp-stream: postpone the creation of the UDP sources
4985 Code refactoring: allocate the UDP ports after the sender and
4986 the reciver parts have been created.
4987 We postpone the creation of the UDP sources until the UDP
4988 ports have been allocated.
4989 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4991 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
4993 * gst/rtsp-server/rtsp-stream.c:
4994 rtsp-stream: added function for setting UDP sources to PLAYING state
4995 Code refactoring: Introduced a function for setting UDP sources
4997 https://bugzilla.gnome.org/show_bug.cgi?id=757488
4999 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
5001 * gst/rtsp-server/rtsp-stream.c:
5002 rtsp-stream: added function for creating and configuring UDP sources
5003 Code refactoring: create and configure UDP sources in a separate function.
5004 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5006 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
5008 * gst/rtsp-server/rtsp-stream.c:
5009 rtsp-stream: added function for RTP/RTCP socket configuration
5010 Code refactoring: configure RTP and RTCP sockets for UDP sinks
5011 in a separate function.
5012 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5014 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
5016 * gst/rtsp-server/rtsp-stream.c:
5017 rtsp-stream: added function for creating and configuring UDP sinks
5018 Code refactoring: create and configure UDP sinks in a separate function.
5019 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5021 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
5023 * gst/rtsp-server/rtsp-stream.c:
5024 rtsp-stream: added helper function for creating the sender/receiver parts
5025 Code refactoring: introduced helper function for creating
5026 the receiver and the sender parts of the streaming pipeline.
5027 https://bugzilla.gnome.org/show_bug.cgi?id=757488
5029 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
5034 === release 1.7.2 ===
5036 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5042 * gst-rtsp-server.doap:
5045 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
5047 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5048 uninstalled.pc: add support for non libtool build systems
5049 Currently the .la path is provided which requires to use libtool as
5050 mentioned in the GStreamer manual section-helloworld-compilerun.html.
5051 It is fine as long as the application is built using libtool.
5052 So currently it is not possible to compile a GStreamer application
5053 within gst-uninstalled with CMake or other build system different
5055 This patch allows to do the following in gst-uninstalled env:
5056 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
5057 gstreamer-rtsp-server-1.0)
5058 Previously it required to prepend libtool --mode=link
5059 https://bugzilla.gnome.org/show_bug.cgi?id=720778
5061 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5063 * gst/rtsp-sink/gstrtspclientsink.c:
5064 rtspclientsink: remove check for impossible condition
5065 Goto error label checks stream to see if it needs to be unreferenced before
5066 returning, but this goto jumps happens before the stream is ever set, so it
5067 will always be NULL in this error label.
5070 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
5072 * gst/rtsp-sink/gstrtspclientsink.c:
5073 rtspclientsink: clean switch statements
5074 Coverity demands for fallthrough statements to be clearly commented,
5075 to distinguish from accidental fall throughs. And it also needs all
5076 cases to finish with a break, even if the break is never going to be
5077 executed like in the case of a continue jump.
5081 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5083 * tests/check/Makefile.am:
5084 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
5085 To get the CK_DEFAULT_TIMEOUT defined for all tests
5086 Also removes a 120 seconds timeout that was set as default
5087 explicitly in this module
5088 https://bugzilla.gnome.org/show_bug.cgi?id=761472
5090 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
5094 Automatic update of common submodule
5095 From 86e4663 to b64f03f
5097 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
5099 * gst/rtsp-server/rtsp-media.c:
5100 rtsp-media: fix state_lock not locked again when preroll fails
5101 https://bugzilla.gnome.org/show_bug.cgi?id=761399
5103 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
5106 configure: Move plugin specific flags below all the others
5107 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
5108 -no-undefined. And -no-undefined is required on Windows to build DLLs.
5110 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
5112 * gst/rtsp-sink/gstrtspclientsink.c:
5113 rtspclientsink: Simplify slightly using new -base API
5114 Use the new Mikey and SDP API in the base plugins libs
5115 to simplify some code.
5116 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5118 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5123 * gst/rtsp-sink/Makefile.am:
5124 * gst/rtsp-sink/gstrtspclientsink.c:
5125 * gst/rtsp-sink/gstrtspclientsink.h:
5126 * gst/rtsp-sink/plugin.c:
5127 * tests/check/Makefile.am:
5128 * tests/check/gst/rtspclientsink.c:
5129 rtspsink: Add rtspclientsink element
5130 Add an rtspclientsink element that accepts streams for which
5131 there is a registered payloader and sends them to
5132 an RTSP server using RECORD.
5133 Sending is synchronised to the pipeline clock. Payload-types
5134 are automatically selected. The 'new-payloader' signal is fired
5135 for custom configuration of payloaders when they are created.
5136 Can now stream a movie like this:
5138 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
5139 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
5141 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
5142 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
5143 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5145 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5147 * gst/rtsp-server/rtsp-stream.c:
5148 * gst/rtsp-server/rtsp-stream.h:
5149 rtsp-stream: Add functions for using rtsp-stream from the client
5150 Add a boolean to indicate that the rtsp-stream is running on the
5151 'client' side of an RTSP connection, for sending streams via
5152 RECORD. In that case, the roles of the client/server ports
5153 in transport setup are swapped.
5154 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5156 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5158 * gst/rtsp-server/rtsp-sdp.c:
5159 * gst/rtsp-server/rtsp-sdp.h:
5160 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
5161 A new function that adds info from a GstRTSPStream into an SDP message.
5162 https://bugzilla.gnome.org/show_bug.cgi?id=758180
5164 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
5166 * gst/rtsp-server/rtsp-media.c:
5167 rtsp-media: Fix mutex beeing unlocked while they should be locked
5168 https://bugzilla.gnome.org/show_bug.cgi?id=761226
5170 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
5172 * gst/rtsp-server/rtsp-media-factory.c:
5173 rtsp-media-factory: add missing break in "clock" property setter
5176 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
5178 * gst/rtsp-server/rtsp-stream.c:
5179 rtsp-stream: fixed assert during update transport
5180 When RTSP server trying update transport during multicast, it throws an
5181 assert. The assert is thrown because it is trying to get the parent of
5182 an non-existing funnel element.
5183 https://bugzilla.gnome.org/show_bug.cgi?id=760150
5185 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
5187 * gst/rtsp-server/rtsp-permissions.h:
5188 * gst/rtsp-server/rtsp-thread-pool.h:
5189 * gst/rtsp-server/rtsp-token.h:
5190 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
5191 gtk-doc can handle static inline functions just fine these days,
5192 there's no need for this stuff any more.
5194 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5196 * gst/rtsp-server/rtsp-media.c:
5197 * gst/rtsp-server/rtsp-sdp.c:
5198 sdp: replace duplicated codes to call new base sdp apis
5199 https://bugzilla.gnome.org/show_bug.cgi?id=745880
5201 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
5203 * examples/test-netclock.c:
5204 test-netclock: Use the new API to configure a clock directly
5206 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
5208 * gst/rtsp-server/rtsp-media-factory.c:
5209 * gst/rtsp-server/rtsp-media-factory.h:
5210 * gst/rtsp-server/rtsp-media.c:
5211 * gst/rtsp-server/rtsp-media.h:
5212 rtsp-media: Add API to directly configure a clock on the media pipelines
5214 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5216 * gst/rtsp-server/rtsp-media.c:
5217 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
5219 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5221 * gst/rtsp-server/rtsp-media-factory.c:
5222 rtsp-media-factory: Add FIXME for 2.0
5224 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
5226 * gst/rtsp-server/rtsp-stream.c:
5227 rtsp-stream: Fix indentation
5229 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5231 * gst/rtsp-server/rtsp-media.c:
5232 rtsp-media: Do not prepare media after media times out
5233 Deferred calls to start_prepare() can be deferred past the point until
5234 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
5235 prepared to wait. Previously there was no lock and no check for this
5236 situation. This meant that a media could be prepared and unprepared
5237 simultaneously by two different threads. Now a lock is in place and a
5238 suitable check is done.
5239 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
5241 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5243 * gst/rtsp-server/rtsp-client.c:
5244 * gst/rtsp-server/rtsp-media-factory.c:
5245 * gst/rtsp-server/rtsp-media-factory.h:
5246 * gst/rtsp-server/rtsp-media.c:
5247 * gst/rtsp-server/rtsp-media.h:
5248 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
5249 Without TEARDOWN it might be desireable to keep the media running and continue
5250 sending data to the client, even if the RTSP connection itself is
5252 Only do this for session medias that have only UDP transports. If there's at
5253 least on TCP transport, it will stop working and cause problems when the
5254 connection is disconnected.
5255 https://bugzilla.gnome.org/show_bug.cgi?id=758999
5257 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
5262 === release 1.7.1 ===
5264 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
5270 * gst-rtsp-server.doap:
5273 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
5276 configure: Make -Bsymbolic check work with clang.
5277 Update the -Bsymbolic check with the version glib has. This version
5279 https://bugzilla.gnome.org/show_bug.cgi?id=759713
5281 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5283 * gst/rtsp-server/rtsp-session-pool.c:
5284 rtsp-session-pool: Avoid dollar sign ($) in session ids
5285 Live555 in VLC strips off dollar signs and then gets very confused,
5286 we don't loose too much entropy by just skipping it.
5288 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
5290 * gst/rtsp-server/rtsp-address-pool.h:
5291 * gst/rtsp-server/rtsp-auth.h:
5292 * gst/rtsp-server/rtsp-client.h:
5293 * gst/rtsp-server/rtsp-media-factory-uri.h:
5294 * gst/rtsp-server/rtsp-media-factory.h:
5295 * gst/rtsp-server/rtsp-media.h:
5296 * gst/rtsp-server/rtsp-mount-points.h:
5297 * gst/rtsp-server/rtsp-permissions.h:
5298 * gst/rtsp-server/rtsp-server.h:
5299 * gst/rtsp-server/rtsp-session-media.h:
5300 * gst/rtsp-server/rtsp-session-pool.h:
5301 * gst/rtsp-server/rtsp-session.h:
5302 * gst/rtsp-server/rtsp-stream-transport.h:
5303 * gst/rtsp-server/rtsp-stream.h:
5304 * gst/rtsp-server/rtsp-thread-pool.h:
5305 * gst/rtsp-server/rtsp-token.h:
5306 rtsp-server: Add g_autoptr() support to all types
5307 https://bugzilla.gnome.org/show_bug.cgi?id=754464
5309 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
5311 * gst/rtsp-server/rtsp-stream.c:
5312 rtsp-stream: fixed valgrind error
5313 Fixed the valgrind error in unit test. The UDP source created during
5314 gst_rtsp_stream_join_bin() was not released while destroying the rtp
5316 https://bugzilla.gnome.org/show_bug.cgi?id=759010
5318 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5322 Automatic update of common submodule
5323 From b319909 to 86e4663
5325 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
5327 * gst/rtsp-server/rtsp-client.c:
5328 rtsp-client: suspend media during setup request
5329 SETUP request from clients needs to suspend the media to clear the
5330 prerolled buffers. Otherwise it will not affect the prerolled buffer
5331 and the prerolled buffers will be incorrect (for example block-size
5332 from setup request will not affect the prerolled buffer unless the
5333 media is suspended).
5334 https://bugzilla.gnome.org/show_bug.cgi?id=758268
5336 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
5338 * gst/rtsp-server/rtsp-stream.c:
5339 rtsp-stream: create stream pipeline based on transport
5340 Based on the protocol, create the rtsp stream pipeline. If only TCP or
5341 only UDP is set as the transport protocol, it will not add the extra tee
5342 or queue element to the pipeline. Both these elements will be added, if
5343 it supports both TCP and UDP protocols. This improves the pipeline
5344 performance when one protocol is present.
5345 https://bugzilla.gnome.org/show_bug.cgi?id=758179
5347 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
5349 * gst/rtsp-server/rtsp-stream.c:
5350 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
5351 Adding them when not needed will start some logic inside rtpbin that might be
5352 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
5353 would start up a rtpjitterbuffer and behave in weird ways.
5354 We still set up the UDP sources for RTP receiving for a sender media to be
5355 able to receive any packets sent by the client for NAT traversal. They will
5356 all go to a fakesink though.
5357 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
5358 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
5359 receive ASYNC_DONE after a seek.
5360 https://bugzilla.gnome.org/show_bug.cgi?id=758319
5362 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
5364 * gst/rtsp-server/rtsp-stream.c:
5365 rtsp-stream: Disable multicast loopback for the multicast udp sources too
5366 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
5367 Previously we were only setting this for sender sockets, which caused looped
5368 back packets to be received on Windows if a multicast transport was used.
5370 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5372 * examples/test-record-auth.c:
5373 * examples/test-record.c:
5374 examples: Actually use the provided port in the record examples
5376 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5378 * examples/test-record-auth.c:
5379 test-record-auth: Add the option to build in TLS support
5381 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5383 * examples/test-auth.c:
5384 test-auth: Use an 'anonymous' user for unauthenticated default
5385 There's a comment on one of the resources that 'user' and 'admin'
5386 shouldn't even be able to see it, but they can if the default
5387 token is 'admin2', since that gives them access anyway.
5389 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5391 * examples/.gitignore:
5392 * examples/Makefile.am:
5393 * examples/test-record-auth.c:
5394 Add test-record-auth example
5396 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
5398 * gst/rtsp-server/rtsp-client.c:
5399 * tests/check/gst/client.c:
5400 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
5402 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
5404 * gst/rtsp-server/rtsp-server.c:
5405 rtsp-server: Change the logic so we don't pop a NULL context
5406 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
5407 will sometimes fail. This call is made before any context is pushed
5408 resulting in an attempt to pop a NULL context.
5409 https://bugzilla.gnome.org/show_bug.cgi?id=757949
5411 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
5413 * tests/check/gst/rtspserver.c:
5414 rtspserver: Add udp-mcast transport SETUP test
5415 Refactor utility functions in the test file so they can handle
5416 more than UDP and TCP as lower transport.
5417 https://bugzilla.gnome.org/show_bug.cgi?id=756969
5419 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
5421 * gst/rtsp-server/rtsp-stream.c:
5422 rtsp-stream: Always unref return value of gst_object_get_parent()
5423 Fixes a leak of a GstBin in the udp-mcast case.
5424 https://bugzilla.gnome.org/show_bug.cgi?id=756968
5426 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
5429 Automatic update of common submodule
5430 From b99800a to b319909
5432 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
5435 Use new GST_ENABLE_EXTRA_CHECKS #define
5436 https://bugzilla.gnome.org/show_bug.cgi?id=756870
5438 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5441 Automatic update of common submodule
5442 From 6babecd to b99800a
5444 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5447 Update GLib dependency to 2.40.0
5449 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5451 * examples/test-mp4.c:
5452 * gst/rtsp-server/rtsp-stream.c:
5453 stream: listen to sender ssrc signals
5454 https://bugzilla.gnome.org/show_bug.cgi?id=746747
5456 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
5459 common: update for new suppression
5460 Makes check-valgrind pass with glib 2.46
5462 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5464 * gst/rtsp-server/rtsp-media.c:
5465 rtsp-media: Take reference to media that will be prepared
5466 default_prepare() takes a transfer-none reference GstRTSPMedia object.
5467 Later on a g_idle_source_new() is created and a pointer to the media
5468 object is passed as user data. If the media is freed before the idle
5469 source is dispatched the media object pointer is invalid, but the idle
5470 source callback expects it to still be valid. To fix this a reference to
5471 the media object is taken when registering the source callback function
5472 and a corresponding release of the reference is done when the souce is
5474 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
5476 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
5478 * examples/test-launch.c:
5479 * examples/test-mp4.c:
5480 * examples/test-ogg.c:
5481 * examples/test-record.c:
5482 * examples/test-uri.c:
5483 rtsp-server: Fix memory leaks when context parse fails
5484 When g_option_context_parse fails, context and error variables are not getting free'd
5485 which results in memory leaks. Free'ing the same.
5486 And replacing g_error_free with g_clear_error, which checks if the error being passed
5487 is not NULL and sets the variable to NULL on free'ing.
5488 https://bugzilla.gnome.org/show_bug.cgi?id=753863
5490 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
5495 === release 1.6.0 ===
5497 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5503 * gst-rtsp-server.doap:
5506 === release 1.5.91 ===
5508 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
5514 * gst-rtsp-server.doap:
5517 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
5519 * docs/libs/gst-rtsp-server-sections.txt:
5520 * gst/rtsp-server/rtsp-stream.c:
5521 stream: fix docs for recently-added get/set_buffer_size API
5522 https://bugzilla.gnome.org/show_bug.cgi?id=749095
5524 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
5526 * gst/rtsp-server/rtsp-media.c:
5527 rtsp-media: Don't crash on encrypted RTX SDP
5528 In parse_keymgmt(), don't mutate the input string that's been passed
5529 as const, especially since we might need the original value again if
5530 the same key info applies to multiple streams (RTX, for example).
5531 https://bugzilla.gnome.org/show_bug.cgi?id=754753
5533 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
5535 * examples/test-mp4.c:
5536 test-mp4: Support filenames with spaces in them. Error out on too few arguments
5538 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
5540 * examples/test-record.c:
5541 test-record: Check parameter count and print out help
5542 If no launch pipeline was supplied, print out some help
5544 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
5546 * gst/rtsp-server/rtsp-media.c:
5547 * gst/rtsp-server/rtsp-stream.c:
5548 * gst/rtsp-server/rtsp-stream.h:
5549 rtsp-stream: Implement UDP buffer size setting.
5550 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
5552 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
5553 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
5555 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
5557 * gst/rtsp-server/rtsp-media.h:
5558 rtsp-media: Fix small typo causing gtk-doc to complain
5560 === release 1.5.90 ===
5562 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
5568 * gst-rtsp-server.doap:
5571 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5573 * gst/rtsp-server/rtsp-media-factory.c:
5574 media-factory: get port number through gst_rtsp_url_get_port
5575 https://bugzilla.gnome.org/show_bug.cgi?id=753473
5577 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
5579 * tests/check/gst/media.c:
5580 media-test: Removing unnecessary assertion
5581 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5583 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5585 * gst/rtsp-server/rtsp-server.c:
5586 Document that source keeps a ref on server until it's destroyed
5587 https://bugzilla.gnome.org/show_bug.cgi?id=749227
5589 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5591 * tests/check/gst/media.c:
5592 media-test: Test for multiple dynamic payload
5593 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5595 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5597 * gst/rtsp-server/rtsp-media.c:
5598 media: Only add fakesink once per pipeline
5599 The intention is to prevent going PLAYING state before pads are created.
5600 If there was mutilple dynamic payload, it would leak few fakesink and
5601 actually prevent from ever reaching playing state.
5602 https://bugzilla.gnome.org/show_bug.cgi?id=753385
5604 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5606 * gst/rtsp-server/rtsp-media.c:
5607 Revert "rtsp-media: Only add 1 fakesink per pipeline"
5608 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
5610 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5612 * gst/rtsp-server/rtsp-media.c:
5613 rtsp-media: Only add 1 fakesink per pipeline
5614 There should be only one fakesink per pipeline, not per dynpay. This
5615 would lead to element naming clash.
5617 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
5619 * gst/rtsp-server/rtsp-media.c:
5620 rtsp-media: assertion error due to wrong condition check
5621 In media to caps function, reserved_keys array is being used for variable i,
5622 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
5623 changed it to variable j
5624 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5626 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
5628 * gst/rtsp-server/rtsp-media.c:
5629 rtsp-media: Strip keys from the fmtp that we use internally in our caps
5630 Skip keys from the fmtp, which we already use ourselves for the
5631 caps. Some software is adding random things like clock-rate into
5632 the fmtp, and we would otherwise here set a string-typed clock-rate
5633 in the caps... and thus fail to create valid RTP caps
5634 https://bugzilla.gnome.org/show_bug.cgi?id=753009
5636 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5638 * gst/rtsp-server/rtsp-thread-pool.c:
5639 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
5640 https://bugzilla.gnome.org/show_bug.cgi?id=752640
5642 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
5645 Automatic update of common submodule
5646 From f74b2df to 9aed1d7
5648 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
5653 === release 1.5.2 ===
5655 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
5661 * gst-rtsp-server.doap:
5664 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
5666 * gst/rtsp-server/rtsp-client.c:
5667 * gst/rtsp-server/rtsp-client.h:
5668 * tests/check/gst/client.c:
5669 rtsp-client: allow application to decide what requirements are supported
5670 Add "check-requirements" signal and vfunc to allow application
5671 (and subclasses) to check the requirements.
5672 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
5673 https://bugzilla.gnome.org/show_bug.cgi?id=749417
5675 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
5678 Automatic update of common submodule
5679 From 6015d26 to f74b2df
5681 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5683 * gst/rtsp-server/rtsp-media.c:
5684 rtsp-media: Always use real payloader when creating streams
5685 A bin that contains the real payloader might be used as payloader. In this
5686 case we have to get the real payloader for the various properties it provides.
5687 Example use cases for this are bins that payload some media and then have
5688 additional elements that add metadata or RTP extension headers to the stream.
5689 https://bugzilla.gnome.org/show_bug.cgi?id=750800
5691 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5693 * examples/test-netclock-client.c:
5694 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
5696 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
5698 * examples/test-netclock-client.c:
5699 * examples/test-netclock.c:
5700 test-netclock: Use new ntp-time-source property on rtpbin
5701 Select the clock time to be used as NTP time source. This allows proper
5702 synchronization between receivers, independent of sharing base times, and just
5703 requires them to use the same clock.
5705 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5707 * examples/test-netclock-client.c:
5708 * examples/test-netclock.c:
5709 test-netclock: Setting the same base time on sender and receiver is not necessary
5710 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
5712 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5714 * gst/rtsp-server/rtsp-stream.c:
5715 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
5716 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5718 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5720 * docs/libs/gst-rtsp-server.types:
5721 docs: add missing types
5722 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5724 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5726 * docs/libs/gst-rtsp-server-sections.txt:
5727 docs: add missing apis
5728 https://bugzilla.gnome.org/show_bug.cgi?id=750764
5730 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
5732 * examples/test-netclock-client.c:
5733 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
5735 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
5737 * docs/libs/gst-rtsp-server-sections.txt:
5738 * gst/rtsp-server/rtsp-auth.c:
5739 * gst/rtsp-server/rtsp-auth.h:
5740 GstRTSPAuth: Add client certificate authentication support
5741 https://bugzilla.gnome.org/show_bug.cgi?id=750471
5743 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
5745 * examples/test-netclock-client.c:
5746 test-netclock-client: Use new GstClock API to wait for clock synchronization
5748 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
5750 * examples/test-netclock-client.c:
5751 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
5752 A mainloop is needed to get glimagesink to display something on OSX, and
5753 the source-setup signal just makes things a little bit easier.
5755 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
5758 Automatic update of common submodule
5759 From d9a3353 to 6015d26
5761 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
5764 Automatic update of common submodule
5765 From d37af32 to d9a3353
5767 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
5770 Automatic update of common submodule
5771 From 21ba2e5 to d37af32
5773 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
5776 Automatic update of common submodule
5777 From c408583 to 21ba2e5
5779 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
5781 * docs/libs/Makefile.am:
5782 docs: remove variables that we define in the snippet from common
5783 This is syncing our Makefile.am with upstream gtkdoc.
5785 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5788 Automatic update of common submodule
5789 From 44a3517 to c408583
5791 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
5796 === release 1.5.1 ===
5798 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
5804 * gst-rtsp-server.doap:
5807 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
5809 * gst/rtsp-server/rtsp-client.c:
5810 rtsp-client: No flush during Teardown.
5811 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
5812 backlog is empty it can happen that just a part of a message will be
5813 sent and rest is in backlog queue. If then flush during teardown
5814 just a part of message will be sent.This can lead to client miss
5815 teardown response since it expect to get the last part of message.
5816 The flushing during teardown was introduced to fix a deadlock that now
5817 is fixed more generally in handle_request by temporary setting backlog
5819 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
5821 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
5823 * tests/check/Makefile.am:
5824 tests: Use AM_TESTS_ENVIRONMENT
5825 Needed by the new automake test runner and the
5826 current version of the common submodule.
5828 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
5830 * gst/rtsp-server/rtsp-media.h:
5831 * gst/rtsp-server/rtsp-stream.h:
5832 rtsp-server: Use single-include rtsp header to make sure we get all definitions
5834 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
5836 * gst/rtsp-server/rtsp-media.c:
5837 rtsp-media: Mark some more functions static
5839 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
5841 * gst/rtsp-server/rtsp-media.c:
5842 rtsp-media: Only unblock the media in suspend() when actually changing the state
5843 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
5845 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
5847 * examples/test-video-rtx.c:
5848 examples: Use AVPF profile for the RTX example
5850 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
5852 * gst/rtsp-server/rtsp-sdp.c:
5853 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
5855 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5857 * gst/rtsp-server/rtsp-stream.c:
5858 rtsp-stream: get valid clock-rate from last-sample
5859 clock-rate in last-sample's caps is integer, not unsigned.
5860 To get this value properly, variable needs to be type-casted to int.
5861 https://bugzilla.gnome.org/show_bug.cgi?id=747614
5863 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
5867 autogen.sh: only run autopoint if gettext requested in configure.ac
5868 Not just because there happens to be a po directory.
5869 https://bugzilla.gnome.org/show_bug.cgi?id=748058
5871 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
5874 Revert "configure.ac: uncomment gettext version setup"
5875 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
5876 We don't need a gettext setup here and there's no po
5877 directory either, so no reason why autopoint would be
5878 run in the first place.
5879 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
5881 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
5883 * examples/test-multicast.c:
5884 * examples/test-multicast2.c:
5885 * examples/test-sdp.c:
5886 * examples/test-video-rtx.c:
5887 * examples/test-video.c:
5888 * tests/test-cleanup.c:
5889 * tests/test-reuse.c:
5890 Fix timeout function signatures across tests and examples
5892 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
5894 * tests/check/Makefile.am:
5895 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
5896 Make sure the test environment is set up.
5897 https://bugzilla.gnome.org//show_bug.cgi?id=747624
5899 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
5902 configure: bump automake requirement to 1.14 and autoconf to 2.69
5903 This is only required for builds from git, people can still
5904 build tarballs if they only have older autotools.
5905 https://bugzilla.gnome.org//show_bug.cgi?id=747624
5907 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5910 configure.ac: uncomment gettext version setup
5911 Fixes autogen.sh. It would run autopoint, which would complain
5912 that it could not find the gettext version in configure.ac.
5913 https://bugzilla.gnome.org/show_bug.cgi?id=748058
5915 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5917 * examples/test-video-rtx.c:
5918 test-video-rtx: set exact payload type to PCMA payloader
5919 Setting wrong payload type causes failure to do retransmission through audio stream
5920 https://bugzilla.gnome.org/show_bug.cgi?id=747839
5922 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
5924 * gst/rtsp-server/rtsp-media.c:
5925 * gst/rtsp-server/rtsp-stream.c:
5926 * gst/rtsp-server/rtsp-stream.h:
5927 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
5928 Because of duplicated g_signal_connect for request-aux-sender signal,
5929 wrong stream pointer is passed to the signal handler.
5930 Instead of passing each stream, pass stream array and get the relevant stream.
5931 https://bugzilla.gnome.org/show_bug.cgi?id=747839
5933 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
5937 Update autogen.sh to latest version from common
5938 Fixes build after aclocal_check etc. helpers have been removed.
5940 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
5943 Automatic update of common submodule
5944 From bc76a8b to c8fb372
5946 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
5948 * gst/rtsp-server/rtsp-stream.c:
5949 rtsp-stream: Limit the queues to 1 buffer
5950 We only need them to be able to pre-roll, queueing up more data here
5951 is only going to harm latency and memory usage.
5953 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
5955 * gst/rtsp-server/rtsp-stream.c:
5956 rtsp-stream: Update comment and ASCII art to the latest code
5957 We have a queue in front of the udpsink too to prevent the pipeline from
5960 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5962 * gst/rtsp-server/rtsp-stream.c:
5963 rtsp-media: Properly return first rtptime
5964 Instead we where returning first GstBuffer timestamp. This would result
5965 in clock skew and unwanted behaviour in RTSP playback.
5966 https://bugzilla.gnome.org/show_bug.cgi?id=746479
5968 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
5970 * gst/rtsp-server/rtsp-stream.c:
5971 rtsp-stream: Don't leave buffer mapped
5972 If the seq is NULL, the RTP buffer was left mapped. We should always
5975 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
5980 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
5982 * gst/rtsp-server/rtsp-media-factory.c:
5983 * tests/check/gst/client.c:
5984 Fix double semicolons
5986 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
5988 * gst/rtsp-server/rtsp-stream.c:
5989 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
5990 This gives more accurate values than asking the payloader. There might be
5991 queueing happening between the payloader and the sink.
5992 https://bugzilla.gnome.org/show_bug.cgi?id=745704
5994 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
5996 * gst/rtsp-server/rtsp-media.c:
5997 rtsp-media: Don't seek for PLAY if the position will not change
5998 https://bugzilla.gnome.org/show_bug.cgi?id=745704
6000 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
6002 * gst/rtsp-server/rtsp-media.c:
6003 rtsp-media: Don't include payload type in the caps for framesize
6004 When the sdp media attribute framesize are converted to caps
6005 the <payload> should not be included.
6006 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
6007 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
6009 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
6011 * gst/rtsp-server/rtsp-sdp.c:
6012 rtsp-sdp: add payload type to the sdp framesize attribute
6013 The sdp framesize attribute is desribed in RFC6064. It is specified
6014 for payloading of H263 and has the following form
6015 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
6016 should be added to the caps in a payloader and the <payload type> should
6017 be added by the rtsp-server.
6018 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
6020 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6022 * examples/test-uri.c:
6023 examples: test-uri: fix tainted variable
6024 Insignificant but this keeps Coverity happy.
6027 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6029 * examples/.gitignore:
6030 * examples/Makefile.am:
6031 * examples/test-netclock-client.c:
6032 * examples/test-netclock.c:
6033 examples: Add a simple example of network synch for live streams.
6034 An example server and client that works for synchronising live streams
6035 only - as it can't support pause/play.
6037 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
6039 * gst/rtsp-server/rtsp-media-factory.c:
6040 * gst/rtsp-server/rtsp-media-factory.h:
6041 rtsp-media-factory: Add functions to set/get the media gtype
6042 Allow specifying the GType of a GstRtspMedia subclass to create
6043 as a simpler way to get the factory to create a custom
6044 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
6046 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
6048 * gst/rtsp-server/rtsp-media.c:
6049 rtsp-media: fix double unlock in _get_buffer_size()
6050 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
6051 because of double g_mutex_unlock () usage.
6052 https://bugzilla.gnome.org/show_bug.cgi?id=745434
6054 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
6056 * gst/rtsp-server/rtsp-session-pool.c:
6057 * gst/rtsp-server/rtsp-session.c:
6058 * gst/rtsp-server/rtsp-session.h:
6059 rtsp-session: Use monotonic time for RTSP session timeout
6060 Changed RTSP session timeout handling to monotonic time
6061 and deprecating the API for current system time.
6062 This fixes timeouts when the system time changes.
6063 https://bugzilla.gnome.org/show_bug.cgi?id=743346
6065 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
6067 * gst/rtsp-server/rtsp-client.c:
6068 * gst/rtsp-server/rtsp-media.c:
6069 rtsp-client: Only error out in PLAY if seeking actually failed
6070 If the media was just not seekable, we continue from whatever position we are
6071 and let the client decide if that is what is wanted or not.
6072 Only if the actual seek failed, we can't really recover and should error out.
6074 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
6076 * gst/rtsp-server/rtsp-stream.c:
6077 rtsp-stream: Add necessary queues between tee and multiudpsink
6078 https://bugzilla.gnome.org/show_bug.cgi?id=744379
6080 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
6082 * gst/rtsp-server/rtsp-client.c:
6083 * gst/rtsp-server/rtsp-media.c:
6084 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
6085 Instead error out properly the same way as if the SEEKING query already
6088 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
6090 * gst/rtsp-server/rtsp-stream.h:
6091 rtsp-stream: minor code formatting fix
6093 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
6095 * gst/rtsp-server/rtsp-media.c:
6096 rtsp-media: fix logic for collect_streams
6097 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
6098 all streams it knows if it got any, and can check if the transport mode is OK.
6101 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6103 * gst/rtsp-server/rtsp-media.c:
6104 rtsp-media: Don't set the transport mode based on what elements we find
6105 Just print a warning if the one that was set before disagrees with what
6106 elements we found. It must already be set to something before as this
6107 function is called after we received the SDP from ANNOUNCE in RECORD mode,
6108 and we would reject ANNOUNCE if the RECORD flag was not set.
6110 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
6112 * tests/check/gst/rtspserver.c:
6113 tests: rtspserver: rename shadowed variable
6114 We have two different 'sink' variables here,
6115 rename one of them for clarity.
6117 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
6119 * gst/rtsp-server/rtsp-client.c:
6120 rtsp-client: fix awkward if clause
6122 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
6124 * examples/test-uri.c:
6125 examples: test-uri: improve uri argument handling and accept file names
6126 Print an error if the argument passed is not a URI and can't
6127 be converted into one, or no arguments have been provided.
6129 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
6131 * examples/test-uri.c:
6132 examples: test-uri: don't remove mount point after 10 seconds
6133 It's very irritating when trying to test stuff repeatedly
6134 and serves no real purpose other than showing that it can
6137 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
6139 * examples/.gitignore:
6140 examples: add new test-record to .gitignore
6142 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6144 * examples/test-record.c:
6145 * gst/rtsp-server/rtsp-client.c:
6146 * gst/rtsp-server/rtsp-media-factory.c:
6147 * gst/rtsp-server/rtsp-media-factory.h:
6148 * gst/rtsp-server/rtsp-media.c:
6149 * gst/rtsp-server/rtsp-media.h:
6150 * tests/check/gst/rtspserver.c:
6151 rtsp-media: Use flags to distinguish between PLAY and RECORD media
6153 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
6155 * examples/test-record.c:
6156 test-record: Set latency for playback-style example to 2s instead of 200ms
6158 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6160 * tests/check/gst/rtspserver.c:
6161 tests: add some unit tests for ANNOUNCE and RECORD
6162 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6164 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
6166 * gst/rtsp-server/rtsp-client.c:
6167 rtsp-client: fix a couple of leaks in handle_announce
6169 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
6171 * gst/rtsp-server/rtsp-media-factory.c:
6172 * gst/rtsp-server/rtsp-media-factory.h:
6173 * gst/rtsp-server/rtsp-media.c:
6174 * gst/rtsp-server/rtsp-media.h:
6175 rtsp-media: Expose latency setting for setting the rtpbin latency
6177 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6179 * examples/test-record.c:
6180 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
6182 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
6184 * gst/rtsp-server/rtsp-stream.c:
6185 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
6187 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
6189 * examples/Makefile.am:
6190 * examples/test-record.c:
6191 * gst/rtsp-server/rtsp-client.c:
6192 * gst/rtsp-server/rtsp-client.h:
6193 * gst/rtsp-server/rtsp-media-factory.c:
6194 * gst/rtsp-server/rtsp-media-factory.h:
6195 * gst/rtsp-server/rtsp-media.c:
6196 * gst/rtsp-server/rtsp-media.h:
6197 * gst/rtsp-server/rtsp-session-media.c:
6198 * gst/rtsp-server/rtsp-stream.c:
6199 * gst/rtsp-server/rtsp-stream.h:
6200 Add initial support for RECORD
6201 We currently only support media that is RECORD or PLAY only, not both at once.
6202 https://bugzilla.gnome.org/show_bug.cgi?id=743175
6204 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
6206 * gst/rtsp-server/rtsp-stream.c:
6207 rtsp-stream: RTCP and RTP transport cache cookies seperated
6208 RTCP packets were not sent because the same tr_cache_cookie was used for
6209 both RTP and RTCP. So only one of the tr_cache lists were populated
6210 depending on which one was sent first. If the tr_cache list is not
6211 populated then no packets can be sent. Most often this happened to be
6212 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
6213 resulted in both the tr_cache_lists to be populated regardless of which
6215 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
6217 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
6219 * gst/rtsp-server/rtsp-stream.c:
6220 rtsp-stream: fix false compiler warning
6221 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
6223 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
6225 * gst/rtsp-server/rtsp-client.c:
6226 rtsp-client: log interleaved data received
6228 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
6230 * gst/rtsp-server/rtsp-client.c:
6231 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
6233 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6235 * gst/rtsp-server/rtsp-client.c:
6236 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
6238 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6240 * gst/rtsp-server/rtsp-client.c:
6241 rtsp-client: Use a random session ID in the SDP
6242 RFC4566 Section 5.2 says that it should make the username, session id,
6243 nettype, addrtype and unicast address tuple globally unique. Always using
6244 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
6245 Instead let's create a 64 bit random number, which at least brings us
6246 closer to the goal of global uniqueness.
6247 https://tools.ietf.org/html/rfc4566#section-5.2
6249 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
6251 * examples/test-launch.c:
6252 * examples/test-mp4.c:
6253 * examples/test-ogg.c:
6254 * examples/test-uri.c:
6255 examples: Don't call gst_init() and gst_get_option_group()
6256 The latter calls the former at the appropriate time.
6258 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
6260 * gst/rtsp-server/rtsp-client.c:
6261 rtsp-client: Drop trailing \0 of RTSP DATA messages
6262 We add a trailing \0 in GstRTSPConnection to make parsing of
6263 string message bodies easier (e.g. the SDP from DESCRIBE) but
6264 for actual data this means we have to drop it or otherwise
6265 create invalid data.
6267 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
6269 * gst/rtsp-server/rtsp-stream.c:
6270 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
6271 Fixes crash when two threads access handle_new_sample() at the same
6272 time, one for RTP, one for RTCP.
6273 Otherwise, when iterating over the transports cache, it might be modified by
6274 another thread at the same time if the transports cookie has changed.
6275 https://bugzilla.gnome.org/show_bug.cgi?id=742954
6277 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
6279 * gst/rtsp-server/rtsp-stream.c:
6280 rtsp-stream: Set format=TIME on our app sources for TCP
6282 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
6284 * gst/rtsp-server/rtsp-session-pool.c:
6285 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
6286 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
6287 RFC 2326 states that session IDs may consist of alphanumeric as well as
6288 the safe characters $-_.+ -- N.B. the percent character is not allowed.
6289 Previously the session ID was URI-escaped, this meant that any character
6290 which was not alphanumeric or any of the characters +-._~ would be
6291 percent encoded. While the RFC (surprisingly) mentions that linear white
6292 space in session IDs should be URI-escaped, it does not say anything
6293 about other characters. Moreover no white space is allowed in the
6294 session ID. Finally the percent character which is the result of
6295 URI-escaping is not allowed in a session ID.
6296 So there is no reason to do any URI-escaping, and now it is removed.
6297 https://bugzilla.gnome.org/show_bug.cgi?id=742869
6299 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
6302 Automatic update of common submodule
6303 From f2c6b95 to bc76a8b
6305 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
6308 Fix 'make check' from top-level directory
6310 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
6312 * examples/test-launch.c:
6313 * examples/test-mp4.c:
6314 * examples/test-ogg.c:
6315 * examples/test-uri.c:
6316 examples: Add command-line parsing and take a 'port' argument
6317 This allows users to run multiple servers on different ports for testing.
6318 Only done for examples that actually take arguments and hence are capable of
6319 outputting different streams for each instance on each port.
6320 https://bugzilla.gnome.org/show_bug.cgi?id=742115
6322 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
6324 * gst/rtsp-server/rtsp-client.c:
6325 * gst/rtsp-server/rtsp-client.h:
6326 rtsp-client: Add a send_message default signal handler
6327 This allows subclasses to easily hook into the response sending
6328 mechanism without doing everything from a signal, which seems
6329 awkward from subclasses.
6331 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
6334 Automatic update of common submodule
6335 From ef1ffdc to f2c6b95
6337 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6341 configure: add --disable-examples switch
6342 https://bugzilla.gnome.org/show_bug.cgi?id=741678
6344 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
6346 * examples/.gitignore:
6347 * examples/Makefile.am:
6348 * examples/test-video-rtx.c:
6349 examples: add a retransmisison example implementing RFC4588
6350 Currently only SSRC-multiplexed rtx streams are supported
6352 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
6354 * gst/rtsp-server/rtsp-stream.c:
6355 rtsp-stream: Fix some minor memory leaks
6357 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
6359 * gst/rtsp-server/rtsp-media.c:
6360 rtsp-media: Some minor cleanup
6362 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
6364 * gst/rtsp-server/rtsp-stream.c:
6365 rtsp-stream: Fix compiler warnings
6366 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
6367 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6369 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
6370 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
6373 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
6375 * docs/libs/gst-rtsp-server-sections.txt:
6376 * gst/rtsp-server/rtsp-media-factory.c:
6377 * gst/rtsp-server/rtsp-media-factory.h:
6378 * gst/rtsp-server/rtsp-media.c:
6379 * gst/rtsp-server/rtsp-media.h:
6380 * gst/rtsp-server/rtsp-sdp.c:
6381 * gst/rtsp-server/rtsp-stream.c:
6382 * gst/rtsp-server/rtsp-stream.h:
6383 media: implement ssrc-multiplexed retransmission support
6384 based off RFC 4588 and the server-rtpaux example in -good
6386 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
6388 * gst/rtsp-server/rtsp-client.c:
6389 * gst/rtsp-server/rtsp-stream-transport.c:
6390 * gst/rtsp-server/rtsp-stream.c:
6391 rtsp: Ref transports in hash table.
6392 Also ref streams for transports.
6393 This solves a crash when reciving a rtcp after teardown but before
6395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
6397 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
6400 Automatic update of common submodule
6401 From 7bb2bce to ef1ffdc
6403 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
6405 * gst/rtsp-server/rtsp-client.c:
6406 client: refactor cleanup of cached media
6408 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
6410 * tests/check/gst/client.c:
6412 The session leak is now fixed, lets remove those FIXME comments.
6414 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
6416 * tests/check/gst/rtspserver.c:
6417 tests: Test to setup two sessions on one connection
6418 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6420 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
6422 * tests/check/gst/rtspserver.c:
6423 tests: Test setup with tcp transport
6424 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6426 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
6428 * gst/rtsp-server/rtsp-client.c:
6429 client: Configure transport after creating session media
6430 The default implementation of configure_client_transport() in
6431 rtsp-client uses the session media when it chooses channels for
6432 interleaved traffic.
6433 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6435 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
6437 * gst/rtsp-server/rtsp-client.c:
6438 * gst/rtsp-server/rtsp-session-media.c:
6439 client: Stop caching media in client when doing setup
6440 If the media has been managed by a session media, it should not be
6441 cached in the client any longer. The GstRTSPSessionMedia object is now
6442 responsible for unpreparing the GstRTSPMedia object using
6443 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
6445 https://bugzilla.gnome.org/show_bug.cgi?id=739112
6447 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6449 * gst/rtsp-server/rtsp-stream.c:
6450 rtsp-stream: unref srtp decoder when leaving bin
6451 https://bugzilla.gnome.org/show_bug.cgi?id=739481
6453 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6455 * gst/rtsp-server/rtsp-client.c:
6456 rtsp-client: mikey memory leaks
6457 https://bugzilla.gnome.org/show_bug.cgi?id=739383
6459 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
6462 Automatic update of common submodule
6463 From 84d06cd to 7bb2bce
6465 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
6468 Parallelise 'make check-valgrind'
6470 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
6473 Automatic update of common submodule
6474 From a8c8939 to 84d06cd
6476 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
6479 Automatic update of common submodule
6480 From 36388a1 to a8c8939
6482 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6484 * gst/rtsp-server/rtsp-media.c:
6485 rtsp-media: deactivate media when shutting down from paused
6486 This was only done when going directly from playing.
6487 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
6489 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6491 * gst/rtsp-server/rtsp-client.c:
6492 * gst/rtsp-server/rtsp-context.h:
6493 rtsp-client: add stream transport to context
6494 We add the stream transport to the context so we can get the configured
6495 client stream transport in the setup request signal.
6496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
6498 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6500 * gst/rtsp-server/rtsp-stream.c:
6501 stream: release lock even not all transports have been removed
6502 We don't want to keep the lock even we return FALSE because not all the
6503 transports have been removed. This could lead into a deadlock.
6504 https://bugzilla.gnome.org/show_bug.cgi?id=737797
6506 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
6508 * gst/rtsp-server/rtsp-sdp.c:
6509 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
6510 These were renamed in GstRTPBasePayload in 1.0
6512 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6514 * gst/rtsp-server/rtsp-client.c:
6515 client: set session media to NULL without the lock
6516 We need to set session medias to NULL without the client lock otherwise
6517 we can end up in a deadlock if another thread is waiting for the lock
6518 and media unprepare is also waiting for that thread to end.
6519 https://bugzilla.gnome.org/show_bug.cgi?id=737690
6521 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
6523 * gst/rtsp-server/rtsp-media.c:
6524 rtsp-media: Set state to UNPREPARING in all cases
6526 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
6528 * gst/rtsp-server/rtsp-media.c:
6529 media: set state to unpreparing when unprepare is initiated
6530 https://bugzilla.gnome.org/show_bug.cgi?id=737675
6532 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
6534 * gst/rtsp-server/rtsp-client.c:
6535 rtsp-client: Remove backlog limit while processings requests
6536 If the backlog limit is kept two cases of deadlocks may be
6537 encountered when streaming over TCP. Without the backlog
6538 limit this deadlocks can not happen, at the expence of
6540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
6542 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
6544 * gst/rtsp-server/rtsp-client.c:
6545 rtsp-client: do not free main context before rtsp watch
6546 https://bugzilla.gnome.org/show_bug.cgi?id=737110
6548 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
6550 * tests/check/gst/rtspserver.c:
6551 tests: Extend unit test timeout to accomodate for valgrind
6552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6554 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
6556 * gst/rtsp-server/rtsp-client.c:
6557 * gst/rtsp-server/rtsp-session.c:
6558 * gst/rtsp-server/rtsp-stream-transport.c:
6559 rtsp-*: Treat sending packets to clients as keepalive
6560 As long as gst-rtsp-server can successfully send RTP/RTCP data to
6561 clients then the client must be reading. This change makes the server
6562 timeout the connection if the client stops reading.
6563 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6565 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
6567 * gst/rtsp-server/rtsp-client.c:
6568 rtsp-client: Allow backlog to grow while expiring session
6569 Allow the send backlog in the RTSP watch to grow to unlimited size while
6570 attempting to bring the media pipeline to NULL due to a session
6571 expiring. Without this change the appsink element cannot change state
6572 because it is blocked while rendering data in the new_sample callback.
6573 This callback will block until it has successfully put the data into the
6574 send backlog. There is a chance that the send backlog is full at this
6575 point which means that the callback may block for a long time, possibly
6576 forever. Therefore the media pipeline may also be prevented from
6577 changing state for a long time.
6578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
6580 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
6582 * gst/rtsp-server/rtsp-client.c:
6583 rtsp-client: Make old compilers happy
6584 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
6585 Just in case that guint8 doesn't fit in a pointer. Just in case ...
6587 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
6589 * gst/rtsp-server/rtsp-client.c:
6590 client: raise the backlog limits before pausing
6591 We need to raise the backlog limits before pausing the pipeline or else
6592 the appsink might be blocking in the render method in wait_backlog() and
6593 we would deadlock waiting for paused.
6594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
6596 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
6598 * gst/rtsp-server/rtsp-client.c:
6599 client: make define for the WATCH_BACKLOG
6600 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
6602 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
6604 * gst/rtsp-server/rtsp-client.c:
6605 client: simplify session transport handling
6606 link/unlink of the transport in a session was done to keep track of all
6607 TCP transports and to send RTP/RTCP data to the streams. We can simplify
6608 that by putting all the TCP transports in a hashtable indexed with the
6610 We also don't need to link/unlink the transports when we pause/resume
6611 the streams. The same effect is already achieved when we pause/play the
6612 media. Indeed, when we pause the media, the transport is removed from
6613 the media and the callbacks will not be called anymore.
6614 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
6616 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
6618 * gst/rtsp-server/rtsp-stream-transport.c:
6619 * gst/rtsp-server/rtsp-stream-transport.h:
6620 stream-transport: make method to handle received data
6621 Make a method to handle the data received on a channel. It sends the
6622 data to the stream of the transport on the RTP or RTCP pads based on
6625 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
6627 * examples/test-mp4.c:
6628 test: add example of dumping RTCP reports
6630 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
6632 * gst/rtsp-server/rtsp-media.c:
6633 * gst/rtsp-server/rtsp-stream.c:
6634 * gst/rtsp-server/rtsp-stream.h:
6635 rtsp-media: Make sure that sequence numbers are monotonic after pause
6636 The sequence number is not monotonic for RTP packets after pause. The
6637 reason is basepayloader generates a randon sequence number when the
6638 pipeline goes from ready to pause. With this fix generation of sequence
6639 number will be monotonic when going from pause to play request.
6640 https://bugzilla.gnome.org/show_bug.cgi?id=736017
6642 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
6644 * gst/rtsp-server/rtsp-client.c:
6645 rtsp-client: Protect saved clients watch with a mutex
6646 Fixes a crash when close() is called while merging clients
6647 in handle_tunnel(). In that case close() would destroy the
6648 watch while it is still being used in handle_tunnel().
6649 https://bugzilla.gnome.org/show_bug.cgi?id=735570
6651 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
6653 * gst/rtsp-server/rtsp-stream.c:
6654 rtsp-stream: Remove the multicast group udp sources when removing from the bin
6656 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
6658 * gst/rtsp-server/rtsp-media.c:
6659 * gst/rtsp-server/rtsp-stream.c:
6660 * gst/rtsp-server/rtsp-stream.h:
6661 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
6662 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
6663 seeking and will always continue counting the time. This leads to
6664 the NPT after a backwards seek to be something completely different
6665 to the actual seek position.
6666 https://bugzilla.gnome.org/show_bug.cgi?id=732644
6668 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
6670 * examples/test-appsrc.c:
6671 examples: fix another reference leak
6672 gst_rtsp_media_get_element() returns a new ref.
6674 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6676 * examples/test-appsrc.c:
6677 examples: unref element after usage
6678 gst_bin_get_by_name_recurse_up() returns an element
6679 reference that must be unreffed after usage.
6680 https://bugzilla.gnome.org/show_bug.cgi?id=734546
6682 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
6684 * gst/rtsp-server/rtsp-media.c:
6685 signals: Fix copy-pasto in target-state signal offset
6687 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
6691 Makefile: Add usage of build-checks step
6692 Allows building checks without running them
6694 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
6696 * gst/rtsp-server/rtsp-stream.c:
6697 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
6698 When a UDP multicast transport is used it is expected that the server listens
6699 for RTP and RTCP packets on the multicast group with the corresponding port.
6700 Without this we will never get RTCP packets from clients in multicast mode.
6701 https://bugzilla.gnome.org/show_bug.cgi?id=732238
6703 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
6708 === release 1.4.0 ===
6710 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6716 * gst-rtsp-server.doap:
6719 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
6721 * gst/rtsp-server/rtsp-media.h:
6722 media: correct misspelled words in description
6723 https://bugzilla.gnome.org/show_bug.cgi?id=733244
6725 === release 1.3.91 ===
6727 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
6733 * gst-rtsp-server.doap:
6736 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
6738 * docs/libs/gst-rtsp-server-sections.txt:
6741 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
6743 * gst/rtsp-server/rtsp-server.c:
6744 server: implement client REMOVE filter
6746 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
6748 * gst/rtsp-server/rtsp-client.c:
6749 * gst/rtsp-server/rtsp-client.h:
6750 client: expose _close() method
6751 Expose a previously internal close method to close the client
6754 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
6756 * gst/rtsp-server/rtsp-session-pool.c:
6757 session-pool: signal session-removed outside of the lock
6758 Release the lock before emiting the session-removed signal.
6760 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
6762 * gst/rtsp-server/rtsp-client.c:
6763 * gst/rtsp-server/rtsp-server.c:
6764 * gst/rtsp-server/rtsp-session-pool.c:
6765 * gst/rtsp-server/rtsp-session.c:
6766 * gst/rtsp-server/rtsp-stream.c:
6767 filter: Release lock in filter functions
6768 Release the object lock before calling the filter functions. We need to
6769 keep a cookie to detect when the list changed during the filter
6770 callback. We also keep a hashtable to make sure we only call the filter
6771 function once for each object in case of concurrent modification.
6772 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
6774 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
6776 * gst/rtsp-server/rtsp-client.c:
6777 client: check if watch is set in handle_teardown()
6778 The unit tests run without a watch
6780 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6782 * tests/check/gst/client.c:
6783 client tests: send teardown to cleanup session
6785 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
6787 * tests/check/gst/rtspserver.c:
6788 server tests: send teardown to cleanup session
6790 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6792 * gst/rtsp-server/rtsp-client.c:
6793 client: keep ref to client for the session removed handler
6794 This extra ref will be dropped when all client sessions have been
6795 removed. A session is removed when a client sends teardown, closes its
6796 endpoint of the TCP connection or the sessions expires.
6797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6799 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
6801 * gst/rtsp-server/rtsp-client.c:
6802 * gst/rtsp-server/rtsp-session.c:
6803 * tests/check/gst/client.c:
6804 client: manage media in session as a last step
6805 Once we manage a media in a session, we can't unmanage it anymore
6806 without destroying it. Therefore, first check everything before we
6807 manage the media, otherwise if something is wrong we have no way to
6809 If we created a new session and something went wrong, remove the session
6810 again. Fixes a leak in the unit test.
6812 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
6814 * examples/test-mp4.c:
6815 * examples/test-ogg.c:
6816 examples: print 'stream ready at url' for mp4 and ogg example
6818 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
6820 * gst/rtsp-server/rtsp-client.c:
6821 * gst/rtsp-server/rtsp-sdp.c:
6822 rtsp: fix for MIKEY api change
6824 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
6826 * gst/rtsp-server/rtsp-client.c:
6827 client: free watch context only once
6828 The watch context is freed when the source is destroyed. Avoids
6829 a CRITICAL when we try to unref the context twice.
6831 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
6833 * gst/rtsp-server/rtsp-client.c:
6836 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
6838 * gst/rtsp-server/rtsp-client.c:
6839 client: protect sessions with lock
6840 Protect the list of sessions with the lock.
6841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
6843 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
6845 * gst/rtsp-server/rtsp-client.c:
6846 Client: keep a ref to the session
6847 Don't just keep a weak ref to the session objects but use a hard ref. We
6848 will be notified when a session is removed from the pool (expired) with
6849 the new session-removed signal.
6850 Don't automatically close the RTSP connection when all the sessions of
6851 a client are removed, a client can continue to operate and it can create
6852 a new session if it wants. If you want to remove the client from the
6853 server, you have to use gst_rtsp_server_client_filter() now.
6854 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
6855 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
6857 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
6859 * gst/rtsp-server/rtsp-session-pool.c:
6860 * gst/rtsp-server/rtsp-session-pool.h:
6861 session-pool: add session-removed signal
6862 Add a signal to be notified when a session is removed from the pool.
6864 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
6866 * gst/rtsp-server/Makefile.am:
6867 * gst/rtsp-server/rtsp-server.h:
6868 Make rtsp-server.h a single-include header, use it for G-I
6869 https://bugzilla.gnome.org/show_bug.cgi?id=732411
6871 === release 1.3.90 ===
6873 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
6879 * gst-rtsp-server.doap:
6882 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
6884 * gst/rtsp-server/rtsp-stream.c:
6885 stream: crypto can be NULL
6887 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
6889 * gst/rtsp-server/rtsp-client.c:
6890 * gst/rtsp-server/rtsp-media.c:
6891 * gst/rtsp-server/rtsp-mount-points.c:
6892 introspection: add missing allow-none annotations
6893 https://bugzilla.gnome.org/show_bug.cgi?id=730952
6895 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
6897 * gst/rtsp-server/rtsp-address-pool.c:
6898 * gst/rtsp-server/rtsp-media.c:
6899 * gst/rtsp-server/rtsp-session-media.c:
6900 * gst/rtsp-server/rtsp-session-pool.c:
6901 * gst/rtsp-server/rtsp-stream-transport.c:
6902 * gst/rtsp-server/rtsp-stream.c:
6903 * gst/rtsp-server/rtsp-token.c:
6904 introspection: add (nullable) annotations to return values
6905 https://bugzilla.gnome.org/show_bug.cgi?id=730952
6907 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
6909 * gst/rtsp-server/rtsp-client.c:
6910 * gst/rtsp-server/rtsp-stream.c:
6911 gi: improve annotations
6912 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
6914 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
6916 * gst/rtsp-server/rtsp-client.c:
6917 * gst/rtsp-server/rtsp-media-factory.c:
6918 * gst/rtsp-server/rtsp-media.c:
6919 * gst/rtsp-server/rtsp-server.c:
6920 signals: use generic marshal function
6921 Use the generic C marshal function.
6922 Use more explicit type instead of G_TYPE_POINTER
6924 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
6926 * gst/rtsp-server/rtsp-context.h:
6927 context: add type macro
6929 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
6931 * gst/rtsp-server/rtsp-client.c:
6932 * gst/rtsp-server/rtsp-sdp.c:
6933 * gst/rtsp-server/rtsp-sdp.h:
6934 sdp: hide key length defines
6935 They don't have a namespace.
6937 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
6942 === release 1.3.3 ===
6944 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
6950 * gst-rtsp-server.doap:
6953 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
6955 * gst/rtsp-server/rtsp-client.c:
6956 * gst/rtsp-server/rtsp-sdp.c:
6957 * gst/rtsp-server/rtsp-sdp.h:
6958 mikey: add different key length parameters
6959 Add encryption and authentication key length parameters to MIKEY. For
6960 the encoders, the key lengths are obtained from the cipher and auth
6961 algorithms set in the caps. For the decoders, they are obtained while
6962 parsing the key management from the client.
6963 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
6965 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
6967 * tests/check/gst/stream.c:
6968 stream tests: Make sure we get right multicast address from stream
6969 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
6971 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
6973 * gst/rtsp-server/rtsp-client.c:
6974 client: ref the context until rtsp watch is alive
6975 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
6977 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
6979 * gst/rtsp-server/rtsp-client.c:
6980 client: Destroy the rtsp watch after connection close
6982 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
6984 * gst/rtsp-server/rtsp-media.c:
6985 media: fix confusing comment
6987 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
6989 * gst/rtsp-server/rtsp-session.c:
6990 rtsp-session: Timeout in header.
6991 Adding the possbilty to always have timout in header.
6992 This is configurabe with setting "timeout-always-visible".
6993 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
6995 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
7000 === release 1.3.2 ===
7002 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
7009 * gst-rtsp-server.doap:
7012 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
7015 Automatic update of common submodule
7016 From 211fa5f to 1f5d3c3
7018 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
7020 * gst/rtsp-server/rtsp-client.c:
7021 client: store TCP ports in transport
7022 Store the TCP ports in the transport when we are doing RTSP over TCP.
7023 This way, we can easily get to the ports from the transport.
7024 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
7026 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7028 * gst/rtsp-server/rtsp-stream.c:
7029 stream: add signals for new RTP/RTCP encoders
7030 New signals to allow the user to configure the dynamically created
7032 https://bugzilla.gnome.org/show_bug.cgi?id=730228
7034 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7036 * gst/rtsp-server/rtsp-media.c:
7037 * gst/rtsp-server/rtsp-media.h:
7038 media: Make suspend()/unsuspend() virtual
7039 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
7041 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
7043 * gst/rtsp-server/rtsp-client.c:
7044 client: fix send-message signal marshaller
7045 Use generic marshalling for the send-message signal. It has
7046 two POINTER arguments, not just one.
7047 https://bugzilla.gnome.org/show_bug.cgi?id=729900
7049 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
7051 * tests/check/gst/media.c:
7052 tests: add and remove pads only once
7053 In this test we simulate a dynamic pad by watching the caps event.
7054 Because of renegotiation in the base payloader now, this caps is sent
7055 multiple times but we can only deal with 1 invocation, use a variable to
7056 only 'add and remove' the pad once.
7058 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
7060 * tests/check/gst/rtspserver.c:
7061 tests: add unit test for correct handling of Require headers
7062 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7064 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7066 * gst/rtsp-server/rtsp-client.c:
7067 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
7068 Servers must handle Require headers and must report a failure
7069 if they don't handle any of the Required options, see RFC 2326,
7070 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
7071 https://bugzilla.gnome.org/show_bug.cgi?id=729426
7073 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
7078 === release 1.3.1 ===
7080 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
7086 * gst-rtsp-server.doap:
7089 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
7092 Automatic update of common submodule
7093 From bcb1518 to 211fa5f
7095 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
7100 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
7102 * tests/check/gst/sessionmedia.c:
7103 tests: fix memory leak in sessionmedia unit test
7105 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
7107 * gst/rtsp-server/rtsp-client.c:
7108 client: emit a signal before sending a message
7109 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
7111 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
7113 * gst/rtsp-server/rtsp-client.c:
7114 client: pass context to send_message
7115 Pass the current context to send_message, we will need it later.
7117 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
7119 * gst/rtsp-server/rtsp-client.c:
7120 client: fix typo in comment
7122 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
7124 * gst/rtsp-server/rtsp-media.c:
7125 media: Do not stop thread twice if default_prepare() fails
7127 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
7129 * gst/rtsp-server/rtsp-client.c:
7130 client: set the watch to flushing before going to NULL
7131 First set the watch to flushing so that we unblock any current and
7132 future attempt to send data on the watch, Then set the pipeline to
7134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
7136 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
7138 * gst/rtsp-server/rtsp-session-pool.c:
7139 * tests/check/gst/sessionpool.c:
7140 rtsp-session-pool: Fixes annotation
7141 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
7142 in the sessionpool test.
7143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
7145 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
7147 * gst/rtsp-server/rtsp-media.c:
7148 * gst/rtsp-server/rtsp-media.h:
7149 media: make media_prepare virtual
7150 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
7152 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7154 * gst/rtsp-server/rtsp-media.c:
7155 * tests/check/gst/media.c:
7156 media: stop the thread in more error cases
7158 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7160 * gst/rtsp-server/rtsp-media.c:
7161 * tests/check/gst/media.c:
7162 media: allow NULL as the thread
7163 Use the default context whan passing a NULL thread.
7165 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7167 * gst/rtsp-server/rtsp-client.c:
7168 rtsp-client: indent cleanup
7169 Coverity was moaning about unreachable code, and I think it was just
7170 confused by { being before the label. We'll see if it pops up again.
7173 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
7175 * gst/rtsp-server/rtsp-client.c:
7176 * gst/rtsp-server/rtsp-media.c:
7177 client: Add drop-backlog property
7178 When we have too many messages queued for a client (currently hardcoded
7179 to 100) we overflow and drop the messages. Add a drop-backlog property
7180 to control this behaviour. Setting this property to FALSE will retry
7181 to send the messages to the client by waiting for more room in the
7183 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
7185 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
7187 * gst/rtsp-server/rtsp-client.c:
7188 client: support for POST before GET when setting up a tunnel
7190 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
7192 * gst/rtsp-server/rtsp-client.c:
7193 client: remove watch of the second client after http tunnel setup
7194 The second client will be freed after the HTTP tunnel has been set up.
7195 Make sure it's RTSP watch is never dispatched again.
7196 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
7198 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
7200 * gst/rtsp-server/rtsp-media.c:
7201 * tests/check/gst/media.c:
7202 media: Make media_prepare() fail if port allocation fails
7203 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
7205 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
7207 * tests/check/gst/media.c:
7208 media test: cleanup the thread pool in tests
7210 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
7212 * gst/rtsp-server/rtsp-media.c:
7213 * tests/check/gst/media.c:
7214 rtsp-media: Unblock blocked streams in unprepare
7215 The streams will be blocked when a live media is prepared.
7216 The streams should be unblocked in gst_rtsp_media_unprepare.
7217 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
7219 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
7221 * gst/rtsp-server/rtsp-media.c:
7222 media: release the state lock when going to NULL
7223 Set our state to UNPREPARING and release the state-lock before
7224 setting the pipeline to the NULL state. This way, any pad-added
7225 callback will be able to take the state-lock and check that we are now
7226 unpreparing instead of deadlocking.
7227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
7229 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
7231 * gst/rtsp-server/rtsp-media.c:
7232 media: protect status with lock
7233 Make sure we only update the status with the lock.
7235 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
7237 * gst/rtsp-server/rtsp-client.c:
7238 * gst/rtsp-server/rtsp-sdp.c:
7239 rtsp: update for MIKEY API changes
7241 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
7243 * gst/rtsp-server/rtsp-client.c:
7244 client: parse the mikey response from the client
7245 Parse the mikey response from the client and update the policy for
7248 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
7250 * gst/rtsp-server/rtsp-stream.c:
7251 * gst/rtsp-server/rtsp-stream.h:
7252 stream: add method to set crypto info
7253 Make a method to configure the crypto information of a stream.
7254 Set udpsrc in READY instead of PAUSED so that we can configure caps
7257 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
7259 * gst/rtsp-server/rtsp-client.c:
7260 client: cleanup error paths
7262 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
7264 * gst/rtsp-server/rtsp-media.c:
7267 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
7269 * examples/test-video.c:
7270 test: enable SRTP only on RTSPS
7271 We only want to enable SRTP when doing rtsp over TLS so that we can
7272 exchange the keys in a secure way.
7274 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
7276 * examples/test-video.c:
7277 test: print an error on failure
7279 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
7282 * examples/test-video.c:
7283 * gst/rtsp-server/rtsp-sdp.c:
7284 * gst/rtsp-server/rtsp-stream.c:
7285 * tests/check/Makefile.am:
7286 stream: add SRTP support
7287 Install srtp encoder and decoder elements in rtpbin
7290 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7292 * tests/check/Makefile.am:
7293 * tests/check/gst/sessionpool.c:
7294 tests: Add unit tests for sessionpool
7295 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
7297 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7299 * tests/check/gst/threadpool.c:
7300 tests: Improve code coverage of rtsp-threadpool tests
7301 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
7303 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7305 * tests/check/gst/sessionmedia.c:
7306 tests: Improve code coverage for rtsp-session-media
7307 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
7309 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7311 gobject-introspection: Add annotations to support language bindings
7312 In addition a few cosmetic changes:
7313 * Adjust the order of arguments
7314 * Fix typo: occured -> occurred
7315 * Fix indentation after Return:-clauses
7316 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
7318 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7320 * gst/rtsp-server/rtsp-stream.c:
7321 rtsp-stream: Don't mix IPv4 and IPv6 addresses
7322 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
7324 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
7326 * gst/rtsp-server/rtsp-stream.c:
7327 stream: take caps after the session manager
7328 Take the caps for the SDP after they leave the rtpbin so that we can
7329 also get the properties added by rtpbin elements.
7331 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
7333 * gst/rtsp-server/rtsp-stream.c:
7334 stream: release lock while pushing out packets
7335 Keep a cache of the transports and use this to iterate the transport
7336 while pushing packets. This allows us to release the lock early.
7337 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
7339 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
7341 * gst/rtsp-server/rtsp-client.c:
7342 * gst/rtsp-server/rtsp-client.h:
7343 rtsp-client: vmethod for modifying tunnel GET response
7344 Add a vmethod tunnel_http_response where the response to the HTTP GET
7345 for tunneled connections can be modified.
7346 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
7348 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
7350 * gst/rtsp-server/rtsp-sdp.c:
7351 sdp: make 1 media line per profile
7352 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
7353 line in the SDP for each profile. The client is then supposed to pick
7354 one of the profiles in the SETUP request. Because the m= lines have the
7355 same pt, the client also knows that only 1 option is possible.
7357 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
7359 * gst/rtsp-server/rtsp-media-factory.c:
7360 * gst/rtsp-server/rtsp-media-factory.h:
7361 * gst/rtsp-server/rtsp-media.c:
7362 factory: add profile property and pass to media and streams
7364 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
7366 * examples/test-multicast.c:
7367 * gst/rtsp-server/rtsp-sdp.c:
7368 sdp: pass multicast connection for multicast-only stream
7369 Pass the multicast address of the stream in the connection info in the
7370 SDP so that clients try a multicast connection first.
7371 Only allow multicast connections in the test-multicast example. Also
7372 increase the TTL a little.
7374 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7377 .gitignore: Ignore gcov intermediate files
7378 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
7380 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
7382 * gst/rtsp-server/rtsp-stream.c:
7383 stream: release some locks in error cases
7385 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7387 docs: Enable and fix gtk-doc warnings
7388 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
7389 * addresspool/mediafactory: Add missing annotation colon
7390 * stream: Annotate return value
7391 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
7393 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
7396 Automatic update of common submodule
7397 From fe1672e to bcb1518
7399 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
7402 Automatic update of common submodule
7403 From 1a07da9 to fe1672e
7405 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
7407 * examples/Makefile.am:
7408 examples: use LDADD for libs instead of LDFLAGS
7410 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
7413 configure: make sure releases are in .doap file
7415 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
7417 * examples/test-cgroups.c:
7418 examples: test-cgroups: don't put code with side effects into g_assert()
7419 The g_assert() might get compiled out with the right
7420 compiler/preprocessor flags.
7422 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
7424 * examples/.gitignore:
7425 examples: add cgroup test binary to .gitignore
7427 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
7429 * examples/test-cgroups.c:
7430 examples: fix cgroup test build
7431 Fixes build failure caused by compiler warning:
7432 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
7434 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
7437 .gitignore: ignore temp files created in the course of 'make check'
7439 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
7441 * gst/rtsp-server/rtsp-media.c:
7442 rtsp-media: don't loose frames handling new PLAY request
7443 If client supplied a range check if the range specifies the start point.
7444 If not, then do an accurate seek to the current position. If a start
7445 point was specified do do a key unit seek to make sure the streaming
7446 starts with decodeable frames.
7447 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
7449 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
7451 * gst/rtsp-server/rtsp-media.c:
7452 Revert "media: only flush when setting a new start position"
7453 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
7454 We need to do the flush in all cases, demuxer block currently for
7457 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
7459 * gst/rtsp-server/rtsp-media.c:
7460 media: only flush when setting a new start position
7461 Only flush the pipeline when we change the start position with
7463 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
7465 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
7467 * gst/rtsp-server/rtsp-stream.c:
7468 stream: set ttl-mc before adding the socket
7469 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
7470 never be set on socket.
7471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
7473 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7475 * gst/rtsp-server/rtsp-media.c:
7476 media: stop thread if media is already prepared
7477 in gst_rtsp_media_prepare() the thread is not used if media is already
7478 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
7480 https://bugzilla.gnome.org/show_bug.cgi?id=724182
7482 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
7485 build: Ship gst-rtsp-server.doap file
7487 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
7489 * tests/check/gst/rtspserver.c:
7490 tests: Fix another compiler warning with gcc
7492 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
7494 * gst/rtsp-server/rtsp-client.c:
7495 * gst/rtsp-server/rtsp-mount-points.c:
7496 * gst/rtsp-server/rtsp-stream.c:
7497 * tests/check/gst/client.c:
7498 rtsp-server: Fix lots of compiler warnings with clang
7500 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
7503 * gst-rtsp-server.doap:
7504 * tests/Makefile.am:
7505 configure: Synchronise with the configure scripts of the other modules
7507 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
7510 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
7512 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
7514 * gst/rtsp-server/rtsp-media.c:
7515 * gst/rtsp-server/rtsp-stream.c:
7516 Revert "rtsp-server: support build against last stable release"
7517 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
7518 Let us require 1.2.3 now, which is going to be released in a few
7521 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
7523 * gst/rtsp-server/rtsp-session-media.c:
7524 * gst/rtsp-server/rtsp-stream-transport.c:
7525 session: improve RTP-Info
7526 Ignore streams that can't generate RTP-Info instead of failing.
7527 Don't return the empty string when all streams are unconfigured but
7528 return NULL so that we don't generate and empty RTP-Info header.
7529 Improve docs a little.
7531 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
7533 * gst/rtsp-server/rtsp-session-media.c:
7534 Don't free rtpinfo GString when it is NULL
7535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7537 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
7539 * gst/rtsp-server/rtsp-media.c:
7540 media: only set keyframe flag when modifying start
7541 Only set the keyframe flag when we modify the start position. The
7542 keyframe flag should probably be ignored when no change is requested but
7543 until we can claim this is all documented properly and all demuxer
7544 implement this, avoid setting the flag.
7545 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
7547 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
7549 * gst/rtsp-server/rtsp-thread-pool.c:
7550 thread-pool: Unref source after mainloop has quit to avoid races in GLib
7551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
7553 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
7555 * gst/rtsp-server/rtsp-stream.c:
7556 stream: handle NULL seqnum and rtptime arguments
7558 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
7560 * gst/rtsp-server/rtsp-thread-pool.c:
7561 * tests/check/gst/threadpool.c:
7562 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
7563 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
7565 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
7567 * gst/rtsp-server/rtsp-stream.c:
7568 stream: add fallback for missing stats property
7569 Use a fallback when the payloader does not have a stats property
7570 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
7572 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
7575 Automatic update of common submodule
7576 From f7bc1c3 to 1a07da9
7578 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
7580 * gst/rtsp-server/rtsp-stream.c:
7581 stream: don't leak stats structure
7582 Don't leak the stats structure and deal with NULL stats.
7584 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
7586 * gst/rtsp-server/rtsp-stream.c:
7587 stream: Get rtpinfo properties atomically from payloader
7588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
7590 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
7592 * gst/rtsp-server/rtsp-media.c:
7593 media: refactor state change functions and signals
7594 Make functions to set the target state and the pipeline state and emit
7595 the signals from those functions.
7597 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
7599 * gst/rtsp-server/rtsp-media.c:
7600 * gst/rtsp-server/rtsp-media.h:
7601 media: add signal to notify of pending state changes
7603 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
7605 * gst/rtsp-server/rtsp-media.c:
7606 * gst/rtsp-server/rtsp-stream.c:
7607 rtsp-server: support build against last stable release
7608 Until 1.2.3 is out with the new get_type function and we
7611 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
7613 * gst/rtsp-server/rtsp-stream.c:
7614 stream: fix compilation
7616 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
7618 * gst/rtsp-server/rtsp-media.c:
7619 * gst/rtsp-server/rtsp-media.h:
7620 * gst/rtsp-server/rtsp-stream.c:
7621 * gst/rtsp-server/rtsp-stream.h:
7622 stream: add property to configure profiles
7624 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
7626 * gst/rtsp-server/rtsp-client.c:
7627 client: let stream check supported transport
7628 Delegate the check if a transport is allowed to the stream.
7629 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
7631 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
7633 * gst/rtsp-server/rtsp-stream.c:
7634 * gst/rtsp-server/rtsp-stream.h:
7635 stream: add method to check supported transport
7636 Add a method to check if a transport is supported
7638 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
7641 configure.ac: Only check for gstreamer-check, not check
7642 We include check in gstreamer-check since quite some time now.
7644 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
7646 * gst/rtsp-server/rtsp-session-media.c:
7647 * gst/rtsp-server/rtsp-stream-transport.c:
7648 * gst/rtsp-server/rtsp-stream.c:
7649 * gst/rtsp-server/rtsp-stream.h:
7650 stream: return clock-rate from get_rtpinfo
7651 And use it to correct the rtptime to the requested start-time.
7652 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
7654 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
7656 * gst/rtsp-server/rtsp-session-media.c:
7657 * gst/rtsp-server/rtsp-stream-transport.c:
7658 * gst/rtsp-server/rtsp-stream-transport.h:
7659 session-media: calculate start-time
7661 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
7663 * gst/rtsp-server/rtsp-stream-transport.c:
7664 * gst/rtsp-server/rtsp-stream.c:
7665 * gst/rtsp-server/rtsp-stream.h:
7666 stream: also return the running-time
7667 Return the running-time in the rtpinfo as well.
7669 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
7671 * gst/rtsp-server/rtsp-client.c:
7672 * gst/rtsp-server/rtsp-session-media.c:
7673 * gst/rtsp-server/rtsp-session-media.h:
7674 * gst/rtsp-server/rtsp-stream-transport.c:
7675 * gst/rtsp-server/rtsp-stream-transport.h:
7676 session-media: let the session-media make the RTPInfo
7677 Add method to create the RTPInfo for a stream-transport.
7678 Add method to create the RTPInfo for all stream-transports in a
7680 Use the session-media RTPInfo code in client. This allows us to refactor
7681 another method to link the TCP callbacks.
7683 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7685 mount-points: sort sequence before g_sequence_lookup
7686 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
7687 sort sequence if dirty, otherwise lookup will fail.
7688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
7690 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
7693 configure: rename package from gst-rtsp to gst-rtsp-server
7694 To match git module name and avoid confusion with the
7695 rtsp lib in gst-plugins-base and rtsp plugin in -good.
7697 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
7700 configure: bump core/base/good requirement to 1.2.0
7701 Bump to released stable version and make implicit
7702 requirements explicit.
7704 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
7709 Fix broken gettext setup which is not used anyway
7711 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
7714 Automatic update of common submodule
7715 From dbedaa0 to d48bed3
7717 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
7719 * gst/rtsp-server/rtsp-client.c:
7720 * gst/rtsp-server/rtsp-media.c:
7721 * gst/rtsp-server/rtsp-media.h:
7722 media: add setup_sdp vmethod
7723 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
7724 gst_rtsp_media_setup_sdp.
7725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
7727 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
7729 * gst/rtsp-server/rtsp-stream.c:
7730 rtsp-stream: Check return value of sscanf
7731 streamid is only valid if sscanf matched something.
7733 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
7735 * gst/rtsp-server/rtsp-client.c:
7736 rtsp-client: Fix iteration
7737 Wouldn't even enter the code block otherwise (i++ was used as the check
7738 and not the postfix).
7740 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
7742 * gst/rtsp-server/rtsp-client.c:
7743 * gst/rtsp-server/rtsp-client.h:
7744 client: add vmethod to configure media and streams
7745 Implement a vmethod that can be used to configure the media and the
7746 streams based on the current context. Handle the blocksize handling in
7747 the default handler.
7748 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
7750 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7753 Make git ignore more unit test binaries
7755 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
7757 * gst/rtsp-server/rtsp-address-pool.h:
7758 * gst/rtsp-server/rtsp-auth.h:
7759 * gst/rtsp-server/rtsp-client.h:
7760 * gst/rtsp-server/rtsp-context.h:
7761 * gst/rtsp-server/rtsp-media-factory-uri.h:
7762 * gst/rtsp-server/rtsp-media-factory.h:
7763 * gst/rtsp-server/rtsp-media.h:
7764 * gst/rtsp-server/rtsp-mount-points.h:
7765 * gst/rtsp-server/rtsp-server.h:
7766 * gst/rtsp-server/rtsp-session-media.h:
7767 * gst/rtsp-server/rtsp-session-pool.h:
7768 * gst/rtsp-server/rtsp-session.h:
7769 * gst/rtsp-server/rtsp-stream-transport.h:
7770 * gst/rtsp-server/rtsp-stream.h:
7771 * gst/rtsp-server/rtsp-thread-pool.h:
7772 * gst/rtsp-server/rtsp-token.h:
7773 rtsp-server: add padding to many public structures
7774 Not mini objects though, since they are not subclassable
7775 anyway, nor kept on the stack or inlined in a structure.
7777 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
7779 media: add new create_rtpbin vmethod
7780 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
7781 https://bugzilla.gnome.org/show_bug.cgi?id=719734
7783 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
7785 * tests/check/gst/media.c:
7786 tests: fix memory leak, free test's thread pool
7787 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
7789 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
7791 * gst/rtsp-server/rtsp-stream-transport.c:
7792 stream-transport: free url in finalize
7794 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
7796 * gst/rtsp-server/rtsp-media.c:
7797 media: also do state change in suspended state
7799 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
7801 * gst/rtsp-server/rtsp-client.c:
7802 * gst/rtsp-server/rtsp-media.c:
7803 media: also handle prepare and range in suspended state
7804 When we are suspended, we are already prepared.
7805 We can get the range in the suspended state.
7807 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
7809 * tests/check/Makefile.am:
7810 * tests/check/gst/sessionmedia.c:
7811 check: add test for uri in setup
7812 Added unit tests for the new functionality in GstRTSPStreamTransport.
7813 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7815 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
7817 * gst/rtsp-server/rtsp-client.c:
7818 client: store setup uri and use in PLAY response
7819 Store the uri used when doing the setup and use that in the PLAY
7821 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
7823 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
7825 * gst/rtsp-server/rtsp-stream-transport.c:
7826 * gst/rtsp-server/rtsp-stream-transport.h:
7827 stream-transport: add method to get/set url
7829 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
7831 * gst/rtsp-server/rtsp-client.c:
7832 client: suspend after SDP and unsuspend before PLAYING
7833 Based on patches by Ognyan Tonchev <ognyan@axis.com>
7834 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
7836 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
7838 * gst/rtsp-server/rtsp-media-factory.c:
7839 * gst/rtsp-server/rtsp-media-factory.h:
7840 * gst/rtsp-server/rtsp-media.c:
7841 * gst/rtsp-server/rtsp-media.h:
7842 * gst/rtsp-server/rtsp-session-media.c:
7843 * gst/rtsp-server/rtsp-session.c:
7844 * tests/check/gst/media.c:
7845 * tests/check/gst/mediafactory.c:
7846 media: add suspend modes
7847 Add support for different suspend modes. The stream is suspended right after
7848 producing the SDP and after PAUSE. Different suspend modes are available that
7849 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
7850 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
7851 state and RESET will bring the pipeline to the NULL state.
7852 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
7853 this means that the pipeline needs to be prerolled again.
7854 Base on patches by Ognyan Tonchev <ognyan@axis.com>
7855 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
7857 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
7859 * gst/rtsp-server/rtsp-media.c:
7860 media: start live streams in blocked state
7861 Start live streams in the blocked state and make them preroll using the
7862 messages. This ensure that no data is played by the sink until we explicitly
7863 unblock the stream right before going to PLAYING.
7864 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
7866 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
7868 * gst/rtsp-server/rtsp-media.c:
7869 media: refactor starting and waiting for preroll
7870 Based on patches from Ognyan Tonchev <ognyan@axis.com>
7871 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
7873 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
7875 * gst/rtsp-server/rtsp-stream.c:
7876 * gst/rtsp-server/rtsp-stream.h:
7877 stream: add API to block streams
7878 Add an API to block on the streams and make it post a message.
7879 Based on patch by Ognyan Tonchev <ognyan@axis.com>
7880 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
7882 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
7884 * docs/libs/Makefile.am:
7885 docs: Specify the override file
7886 Even if it's empty (for now) it avoids make distcheck complaining
7888 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
7890 * gst/rtsp-server/rtsp-media.c:
7891 media: move default implementations to where they are used
7893 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
7895 * gst/rtsp-server/rtsp-media.c:
7896 media: take the right lock in gst_rtsp_media_set_pipeline_state()
7897 We need to take the state_lock when calling this method.
7899 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
7901 * gst/rtsp-server/rtsp-media.c:
7902 media: handle add-added on non-bins too
7903 Handle dynamic payloaders that are not bins, as used in the unit-test.
7905 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7907 * gst/rtsp-server/rtsp-media-factory.c:
7908 * gst/rtsp-server/rtsp-media-factory.h:
7909 * gst/rtsp-server/rtsp-media.c:
7910 rtsp-media/-factory: Fix request pad name comments
7911 These must be escaped for gtk-doc to parse the comments without warnings.
7913 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
7915 rtsp-media: remove transports if media is in error status
7916 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
7917 trying to change to GST_STATE_NULL and media is in error status, we
7918 remove all transports.
7919 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
7921 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
7923 * gst/rtsp-server/rtsp-media.c:
7924 rtsp-media: use element metadata to find payloader
7925 Use the element metadata to find the payloader instead of checking
7927 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
7929 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
7931 rtsp-stream: add getter for payload type
7932 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
7933 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
7934 element and create the stream with this one instead of the dynpay%d
7936 https://bugzilla.gnome.org/show_bug.cgi?id=712396
7938 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7940 * gst/rtsp-server/rtsp-client.c:
7941 * gst/rtsp-server/rtsp-context.h:
7942 * gst/rtsp-server/rtsp-media.c:
7943 * gst/rtsp-server/rtsp-mount-points.c:
7944 * gst/rtsp-server/rtsp-server.c:
7945 * gst/rtsp-server/rtsp-token.c:
7946 rtsp-*: Refer to NULL as a constant in comments
7948 https://bugzilla.gnome.org/show_bug.cgi?id=714988
7950 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7952 rtsp-*: Fix type name typos in comments
7953 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
7954 * rtsp-auth: Refer to part of constant name as text
7955 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
7956 * rtsp-session-media: Fix GstRTSPSessionMedia typo
7957 * rtsp-stream: Fix typo when refering to GstBin
7958 https://bugzilla.gnome.org/show_bug.cgi?id=714988
7960 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7963 * docs/libs/gst-rtsp-server-docs.sgml:
7964 * docs/libs/gst-rtsp-server-sections.txt:
7965 docs: Improve documentation
7966 * Include annotation-glossary to quiet gtk-doc
7967 * Rename remaining ClientState -> Context
7968 * Rename object hierarchy file
7969 * Remove stale chapter references
7970 * Add missing function and object references
7971 * Include missing GstRTSPAddressPoolResult
7972 https://bugzilla.gnome.org/show_bug.cgi?id=714988
7974 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
7976 * gst/rtsp-server/rtsp-client.c:
7977 * gst/rtsp-server/rtsp-server.c:
7978 * gst/rtsp-server/rtsp-session-pool.c:
7979 * gst/rtsp-server/rtsp-session.c:
7980 * gst/rtsp-server/rtsp-stream.c:
7981 rtsp-server: sprinkle some allow-none annotations for g-i
7983 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
7985 * gst/rtsp-server/rtsp-stream.c:
7986 * gst/rtsp-server/rtsp-stream.h:
7987 stream: add method to filter transports
7988 Add a method to safely iterate and collect the stream transports
7989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
7991 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
7993 * gst/rtsp-server/rtsp-client.c:
7994 * gst/rtsp-server/rtsp-server.c:
7995 * gst/rtsp-server/rtsp-session-pool.c:
7996 * gst/rtsp-server/rtsp-session.c:
7997 rtsp: allow NULL func in filters
7998 Passing a null function make the filters return a list of
8001 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
8003 * gst/rtsp-server/rtsp-address-pool.c:
8004 * tests/check/gst/addresspool.c:
8005 address-pool: fix address increment
8006 Use a guint instead of guint8 to increment the address. It's still not
8007 completely correct because a guint might not be able to hold the complete
8008 address range, but that's an enhacement for later.
8009 Add unit test to test improved behaviour.
8010 https://bugzilla.gnome.org/show_bug.cgi?id=708237
8012 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
8014 * gst/rtsp-server/rtsp-client.c:
8015 * tests/check/gst/client.c:
8016 client: allow absolute path in requests
8017 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
8019 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
8021 * gst/rtsp-server/rtsp-client.c:
8022 * gst/rtsp-server/rtsp-client.h:
8023 client: make make_path_from_uri a vmethod
8025 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8027 * docs/libs/gst-rtsp-server-sections.txt:
8028 * gst/rtsp-server/rtsp-stream.c:
8029 * gst/rtsp-server/rtsp-stream.h:
8030 * tests/check/Makefile.am:
8031 * tests/check/gst/stream.c:
8032 stream: Add functions to get rtp and rtcp sockets
8033 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
8035 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8037 * gst/rtsp-server/rtsp-context.c:
8038 * gst/rtsp-server/rtsp-context.h:
8039 context: defing a GType for the context
8040 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
8042 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8044 * gst/rtsp-server/Makefile.am:
8045 * gst/rtsp-server/rtsp-auth.c:
8046 * gst/rtsp-server/rtsp-context.c:
8047 * gst/rtsp-server/rtsp-media.c:
8048 * gst/rtsp-server/rtsp-mount-points.c:
8049 * gst/rtsp-server/rtsp-server.h:
8050 * gst/rtsp-server/rtsp-session-media.c:
8051 * gst/rtsp-server/rtsp-session.c:
8052 * gst/rtsp-server/rtsp-stream.c:
8053 Fixed several GIR warnings
8055 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
8057 * gst/rtsp-server/rtsp-auth.c:
8060 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8062 * tests/check/Makefile.am:
8063 * tests/check/gst/token.c:
8064 tests: Add unit tests for token
8065 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8067 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8069 * gst/rtsp-server/rtsp-token.c:
8070 token: Validate args for gst_rtsp_token_is_allowed
8071 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
8073 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8075 * gst/rtsp-server/rtsp-token.c:
8076 token: Fix bug when creating empty token
8077 We always want to have a valid GstStructure in the token.
8078 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
8080 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
8082 * gst/rtsp-server/rtsp-thread-pool.c:
8083 thread-pool: avoid race in shutdown
8084 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
8085 don't actually stop the mainloop ever. Solve this race by adding an idle source
8086 to the mainloop that calls the _quit. This way we immediately exit the mainloop
8087 if quit was called before we started it.
8089 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8091 * tests/check/Makefile.am:
8092 * tests/check/gst/permissions.c:
8093 tests: Add unit tests for permissions
8094 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
8096 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8098 * tests/check/gst/mediafactory.c:
8099 tests: Test mediafactory permissions
8100 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8102 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8104 * gst/rtsp-server/rtsp-permissions.c:
8105 permissions: Fix refcounting when adding/removing roles
8106 Previously a role that was removed was unreffed twice, and when
8107 replacing an existing role the replaced role was freed while still being
8108 referenced. Both bugs are now fixed.
8109 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8111 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8113 * tests/check/gst/media.c:
8114 * tests/check/gst/mediafactory.c:
8115 * tests/check/gst/rtspserver.c:
8116 tests: Check gst_rtsp_url_parse return value
8117 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
8119 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
8122 Automatic update of common submodule
8123 From 865aa20 to dbedaa0
8125 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
8127 * gst/rtsp-server/rtsp-server.c:
8128 rtsp-server: Fix socket leak
8129 https://bugzilla.gnome.org/show_bug.cgi?id=710088
8131 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
8133 * gst/rtsp-server/rtsp-session-pool.c:
8134 rtsp-session-pool: Make sure session IDs are properly URI-escaped
8135 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8137 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8139 * examples/.gitignore:
8140 * examples/test-video.c:
8141 examples: fix compilation when WITH_AUTH is defined
8142 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8144 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
8147 gitignore: Add new test binary
8149 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
8151 * tests/check/Makefile.am:
8152 * tests/check/gst/threadpool.c:
8153 thread-pool: Add unit test for the thread pools
8154 https://bugzilla.gnome.org/show_bug.cgi?id=710228
8156 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
8158 * gst/rtsp-server/rtsp-thread-pool.c:
8159 thread-pool: Fix thread leak when reusing threads
8160 https://bugzilla.gnome.org/show_bug.cgi?id=709730
8162 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
8164 * gst/rtsp-server/rtsp-server.c:
8165 * tests/check/gst/rtspserver.c:
8166 tests: fixed racy behavior in rtspserver tests
8167 https://bugzilla.gnome.org/show_bug.cgi?id=710078
8169 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
8171 * tests/check/gst/addresspool.c:
8172 tests: Improve address pool unit tests
8173 Add a range with mixed IPV4 and IPV6 addresses to pool.
8174 Get an IPV4 address from an IPV6-only pool.
8175 Get an IPV6 address from an IPV4-only pool.
8176 Reserve a IPV6 address from an IPV4-only pool.
8177 Check for unicast addresses in multicast-only pool.
8178 Check for unicast addresses in uni-/multicast-mixed pool.
8179 https://bugzilla.gnome.org/show_bug.cgi?id=710128
8181 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8183 * gst/rtsp-server/rtsp-client.c:
8184 client: append query string in PAUSE/PLAY/TEARDOWN as well
8186 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
8188 * gst/rtsp-server/rtsp-client.c:
8189 client: Add query to control path
8190 If the SETUP url contains a query it must be appended to the control
8191 path so that it matches any already created stream in the media. The
8192 query will also be appended to the session media path.
8194 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8196 * gst/rtsp-server/rtsp-media.c:
8197 rtsp-media: remove old line
8199 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
8201 * gst/rtsp-server/rtsp-stream.c:
8202 stream: Correct control comparison
8203 https://bugzilla.gnome.org/show_bug.cgi?id=709176
8205 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8207 * gst/rtsp-server/rtsp-media.c:
8208 media: Check dynamically if the pipeline supports seeking
8209 We should not depend on whether or not the pipeline state change
8210 returned NO_PREROLL or not. A media could dynamically change its
8211 element and switch from seekable to non seekable so it's best to test
8212 the seekable nature of the pipeline dynamically when we try to do a seek.
8214 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8216 * gst/rtsp-server/rtsp-media.c:
8217 media: Return FALSE if seeking is not supported
8219 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8221 * gst/rtsp-server/rtsp-media.c:
8222 rtsp-media: don't seek accurate by default
8223 Accurate seeking is perhaps a little overkill in the most common situation and
8224 causes some formats (mp3) over slow media to seek extremely slowly.
8226 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
8228 * tests/check/gst/rtspserver.c:
8229 tests: fix unit test
8230 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
8232 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
8234 * gst/rtsp-server/rtsp-client.c:
8235 client: Reply 400 if media cannot be constructed
8236 Reply 400 Bad Request instead of 503 Service Unavailable if media
8237 cannot be constructed in SETUP.
8238 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
8240 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
8242 * gst/rtsp-server/rtsp-client.c:
8243 client: Send setup reply once only
8244 If find_media() failed in handle_setup_request() two replies was sent.
8245 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
8247 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
8250 Automatic update of common submodule
8251 From 6b03ba7 to 865aa20
8253 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
8255 * gst/rtsp-server/rtsp-server.c:
8256 server: Emit client-connected signal earlier
8257 Emit client-connected before the client ref is given to a GSource,
8258 otherwise client-connected can be emitted after the client object has
8261 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
8263 * gst/rtsp-server/rtsp-address-pool.c:
8264 * gst/rtsp-server/rtsp-address-pool.h:
8265 * gst/rtsp-server/rtsp-stream.c:
8266 * tests/check/gst/addresspool.c:
8267 addresspool: return reason of failure
8268 Let gst_rtsp_address_pool_reserve_address() return the reason why
8269 the address could not be reserved.
8270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
8272 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
8275 autogen.sh: Sync behaviour with other GStreamer modules
8276 Allows building from outside of tree amongst other things
8278 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
8281 Automatic update of common submodule
8282 From b613661 to 6b03ba7
8284 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
8287 Automatic update of common submodule
8288 From 74a6857 to b613661
8290 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
8293 Automatic update of common submodule
8294 From 01a7a46 to 74a6857
8296 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
8298 * gst/rtsp-server/rtsp-client.c:
8299 client: Do not read beyond end of path string
8300 If the setup was done without a control url, make sure we don't try to read the
8301 non-existing control string and crash.
8303 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8305 * gst/rtsp-server/rtsp-client.c:
8306 client: Fix RTPInfo header
8307 Refactor the method to make the content_base.
8308 Use the content-base and the control url to construct the RTPInfo
8311 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8313 * gst/rtsp-server/rtsp-client.c:
8314 client: map url to path only in describe
8315 Only map the request url to a path in the DESCRIBE method. The SDP then
8316 contains the base and control urls that should be used to SETUP/PAUSE/
8317 PLAY/TEARDOWN the media.
8319 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8321 * gst/rtsp-server/rtsp-client.c:
8322 Revert "client: map URL to path in requests"
8323 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
8324 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
8325 contains the base and control urls which are used in the SETUP, PLAY,
8326 PAUSE and TEARDOWN requests.
8328 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8330 * gst/rtsp-server/rtsp-client.c:
8331 client: map URL to path in requests
8333 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8335 * gst/rtsp-server/rtsp-client.c:
8336 * gst/rtsp-server/rtsp-mount-points.c:
8337 * gst/rtsp-server/rtsp-mount-points.h:
8338 mount-points: make vmethod to make path from uri
8339 Make a vmethod to transform an url into a path. The path is then used to lookup
8340 the factory. This makes it possible to also use other bits of the url, such as
8341 the query parameters, to locate the factory.
8343 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
8345 * gst/rtsp-server/rtsp-thread-pool.c:
8346 * gst/rtsp-server/rtsp-thread-pool.h:
8347 thread-pool: Add cleanup to wait for the threadpool to finish
8348 Also fix race condition if two threads are asking for the first
8349 thread from the thread pool at once. This would case two internal
8350 GThreadPools to be created.
8351 https://bugzilla.gnome.org/show_bug.cgi?id=707753
8353 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
8355 * gst/rtsp-server/rtsp-client.c:
8356 * tests/check/gst/client.c:
8357 client: free threadpool
8358 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8360 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
8362 * tests/check/gst/mountpoints.c:
8363 mountpoints tests: unref matched factories
8364 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8366 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
8368 * tests/check/gst/media.c:
8369 media tests: unref thread pool and caps
8370 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8372 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
8374 * gst/rtsp-server/rtsp-auth.c:
8375 * gst/rtsp-server/rtsp-media-factory.c:
8376 * gst/rtsp-server/rtsp-media.c:
8377 auth, media, media-factory: unref permissions
8378 https://bugzilla.gnome.org/show_bug.cgi?id=707638
8380 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8382 * examples/Makefile.am:
8383 Makefile: add rule for appsrc example
8385 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8387 * examples/test-appsrc.c:
8388 tests: add appsrc example
8389 Add an example on how to use appsrc to feed the server pipeline with data.
8391 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
8393 * gst/rtsp-server/rtsp-client.c:
8394 rtsp-client: remove query part from content-base string
8395 Make sure that after the control url has been resolved, it's
8396 not a part of the query-string.
8397 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
8399 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8401 * gst/rtsp-server/rtsp-client.c:
8402 client: don't check url in response
8403 There is no url or method in the response to check
8405 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8407 * gst/rtsp-server/rtsp-client.c:
8408 * gst/rtsp-server/rtsp-client.h:
8409 Add handle-response signal for when we receive a GET_PARAMETER response
8411 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8413 * gst/rtsp-server/rtsp-server.c:
8414 Fix gst_rtsp_server_client_filter, using wrong variable type
8416 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
8418 * gst/rtsp-server/rtsp-media-factory-uri.c:
8419 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
8420 For AAC we need to check for framed=true instead of parsed=true.
8421 https://bugzilla.gnome.org/show_bug.cgi?id=701384
8423 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8425 * gst/rtsp-server/rtsp-stream.c:
8426 stream: optimize pipeline for protocols
8427 When TCP is not an allowed protocol for the stream, avoid creating the
8428 appsrc/appsink/queue and tee elements.
8430 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8432 * gst/rtsp-server/rtsp-media.c:
8433 media: set protocols on streams
8435 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8437 * gst/rtsp-server/rtsp-client.c:
8438 client: use protocols supported by stream
8440 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-media-factory.c:
8443 * gst/rtsp-server/rtsp-media.c:
8444 * gst/rtsp-server/rtsp-stream.c:
8445 media-factory: allow all protocols
8447 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8449 * gst/rtsp-server/rtsp-media.c:
8450 media: configure protocols in new streams
8452 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8454 * gst/rtsp-server/rtsp-stream.c:
8455 * gst/rtsp-server/rtsp-stream.h:
8456 stream: add protocols property
8458 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8460 * gst/rtsp-server/rtsp-media.c:
8461 rtsp-media: send state in "new-state" signal
8462 https://bugzilla.gnome.org/show_bug.cgi?id=705110
8464 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
8467 build: add subdir-objects to AM_INIT_AUTOMAKE
8468 Fixes warnings with automake 1.14
8469 https://bugzilla.gnome.org/show_bug.cgi?id=705350
8471 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8473 * docs/libs/gst-rtsp-server-sections.txt:
8474 * gst/rtsp-server/rtsp-client.c:
8475 * gst/rtsp-server/rtsp-server.c:
8476 * gst/rtsp-server/rtsp-server.h:
8477 server: add method to iterate clients of server
8479 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8481 * gst/rtsp-server/rtsp-media.c:
8482 * gst/rtsp-server/rtsp-media.h:
8483 Add vmethod for rtsp-media subclass to access rtpbin
8485 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8487 * gst/rtsp-server/rtsp-client.h:
8488 small documentation fix
8490 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8492 * gst/rtsp-server/rtsp-client.c:
8493 Do not take range header if range is invalid
8495 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8497 * docs/libs/gst-rtsp-server-sections.txt:
8498 * gst/rtsp-server/rtsp-media.c:
8499 media: add docs for new method
8501 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8503 * gst/rtsp-server/rtsp-media.c:
8504 * gst/rtsp-server/rtsp-media.h:
8505 Add API to rtsp-media set the pipeline's state
8507 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
8509 * gst/rtsp-server/rtsp-media.c:
8510 Update current position/duration when gst_rtsp_media_get_range_string is called
8512 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8514 * examples/test-cgroups.c:
8515 tests: add some more docs
8517 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8519 * examples/test-cgroups.c:
8520 * gst/rtsp-server/Makefile.am:
8521 * gst/rtsp-server/rtsp-auth.c:
8522 * gst/rtsp-server/rtsp-auth.h:
8523 * gst/rtsp-server/rtsp-client.c:
8524 * gst/rtsp-server/rtsp-client.h:
8525 * gst/rtsp-server/rtsp-context.c:
8526 * gst/rtsp-server/rtsp-context.h:
8527 * gst/rtsp-server/rtsp-params.c:
8528 * gst/rtsp-server/rtsp-params.h:
8529 * gst/rtsp-server/rtsp-server.c:
8530 * gst/rtsp-server/rtsp-thread-pool.c:
8531 * gst/rtsp-server/rtsp-thread-pool.h:
8532 * tests/check/gst/client.c:
8533 ClientState -> Context
8534 Rename the clientstate to context and put the code in a separate file.
8536 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8538 * examples/test-auth.c:
8539 * gst/rtsp-server/rtsp-auth.c:
8540 * gst/rtsp-server/rtsp-auth.h:
8541 auth: add support for default token
8542 The default token is used when the user is not authenticated and can be used to
8543 give minimal permissions.
8545 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8547 * examples/test-auth.c:
8548 * gst/rtsp-server/rtsp-auth.c:
8549 auth: use defines when possible
8551 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8553 * gst/rtsp-server/rtsp-address-pool.c:
8554 address-pool: improve docs
8556 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8558 * gst/rtsp-server/rtsp-permissions.c:
8559 permissions: add the role to the copy
8561 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
8563 * gst/rtsp-server/rtsp-permissions.c:
8564 permissions: Also copy the roles
8566 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
8568 * gst/rtsp-server/rtsp-permissions.c:
8569 permissions: Make it build
8571 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8573 * gst/rtsp-server/rtsp-address-pool.h:
8576 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * docs/libs/gst-rtsp-server-sections.txt:
8579 * gst/rtsp-server/rtsp-auth.c:
8580 * gst/rtsp-server/rtsp-auth.h:
8581 * gst/rtsp-server/rtsp-media.c:
8582 * gst/rtsp-server/rtsp-session-media.c:
8583 * gst/rtsp-server/rtsp-stream-transport.c:
8584 * gst/rtsp-server/rtsp-stream-transport.h:
8585 * gst/rtsp-server/rtsp-stream.c:
8586 * tests/check/gst/client.c:
8589 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * docs/libs/gst-rtsp-server-sections.txt:
8592 * gst/rtsp-server/rtsp-address-pool.c:
8593 * gst/rtsp-server/rtsp-address-pool.h:
8594 * tests/check/gst/addresspool.c:
8595 * tests/check/gst/rtspserver.c:
8596 address-pool: cleanups
8597 Remove redundant method, improve docs.
8599 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8601 * docs/libs/gst-rtsp-server-sections.txt:
8602 * gst/rtsp-server/rtsp-auth.h:
8603 * gst/rtsp-server/rtsp-permissions.c:
8604 * gst/rtsp-server/rtsp-permissions.h:
8605 * gst/rtsp-server/rtsp-token.c:
8608 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8610 * gst/rtsp-server/rtsp-permissions.c:
8611 permissions: implement _remove_role
8613 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8615 * gst/rtsp-server/rtsp-permissions.c:
8616 permissions: update docs
8618 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8620 * tests/check/gst/client.c:
8621 tests: simplify tests
8622 Client settings are now disabled by default so we don't need an auth
8623 module to disable them.
8625 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8627 * gst/rtsp-server/rtsp-auth.c:
8628 auth: add default authorizations
8629 When no auth module is specified, use our table of defaults to look up the
8630 default value of the check instead of always allowing everything. This was
8631 we can disallow client settings by default.
8633 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8636 README: update readme
8638 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-thread-pool.c:
8641 * gst/rtsp-server/rtsp-thread-pool.h:
8642 thread-pool: add more docs
8644 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8646 * gst/rtsp-server/rtsp-thread-pool.c:
8647 * gst/rtsp-server/rtsp-thread-pool.h:
8648 thread-pool: fix race in thread reuse
8649 If we try to reuse a thread right after we made it stop, we end up using a
8650 stopped thread. Catch this case and only reuse threads that are not stopping.
8652 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8654 * gst/rtsp-server/rtsp-server.c:
8655 server: add small debug
8657 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * tests/check/gst/client.c:
8661 Add some permissions to media so we can use the auth and enable
8664 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8666 * gst/rtsp-server/rtsp-client.c:
8667 client: support pushed context in handle_request
8668 If we already have a pushed state, reuse it and add our own things. This makes
8669 it easier to write tests.
8671 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8673 * gst/rtsp-server/rtsp-auth.c:
8674 auth: don't auth on methods
8675 Don't authorize on methods anymore but on the resources that we
8676 try to access, this is more flexible.
8677 Move the authorization checks to where they are needed and let the
8678 check return the response on error.
8680 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8682 * gst/rtsp-server/rtsp-mount-points.c:
8683 mount-points: add some debug
8685 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8687 * tests/check/gst/client.c:
8688 tests: almost fix test
8690 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8692 * gst/rtsp-server/rtsp-auth.c:
8693 * gst/rtsp-server/rtsp-auth.h:
8694 * gst/rtsp-server/rtsp-client.c:
8695 * gst/rtsp-server/rtsp-client.h:
8696 * gst/rtsp-server/rtsp-server.c:
8697 * gst/rtsp-server/rtsp-server.h:
8698 auth: let the auth module check client_settings
8699 Let the auth module decide if client settings are allowed for the
8702 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8704 * gst/rtsp-server/rtsp-token.c:
8705 * gst/rtsp-server/rtsp-token.h:
8706 token: add method to check boolean permission
8708 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8710 * examples/test-auth.c:
8711 * examples/test-cgroups.c:
8712 * gst/rtsp-server/rtsp-token.c:
8713 * gst/rtsp-server/rtsp-token.h:
8714 token: simplify token constructor
8715 Use variable arguments to make easier API.
8717 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8719 * examples/test-auth.c:
8720 * examples/test-cgroups.c:
8721 * gst/rtsp-server/rtsp-media-factory.c:
8722 * gst/rtsp-server/rtsp-media-factory.h:
8723 media-factory: add convenience API for factory
8725 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8727 * examples/test-auth.c:
8728 * examples/test-cgroups.c:
8729 * gst/rtsp-server/rtsp-permissions.c:
8730 * gst/rtsp-server/rtsp-permissions.h:
8731 permissions: simplify API a little
8732 Avoid passing GstStructure in the add_role method, use varargs instead
8733 to construct the structure behind the scenes. We can then also use the
8734 structure name as the role and simplify some more logic.
8736 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8738 * gst/rtsp-server/rtsp-auth.c:
8741 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-auth.c:
8744 * gst/rtsp-server/rtsp-auth.h:
8745 * gst/rtsp-server/rtsp-client.c:
8746 auth: handle unauthorized response
8747 Move handling of the unauthorized response to the auth module, it can add
8748 the appropriate headers to request authorization for the required method
8749 much better than the client.
8751 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8753 * gst/rtsp-server/rtsp-client.c:
8754 * gst/rtsp-server/rtsp-client.h:
8755 client: allow for sending any message, not only requests
8756 Change the _send_request() method to _send_message() so that we
8757 can both send requests and replies.
8759 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8761 * docs/libs/gst-rtsp-server-sections.txt:
8762 * gst/rtsp-server/rtsp-server.h:
8765 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8767 * examples/test-video.c:
8768 * gst/rtsp-server/rtsp-auth.c:
8769 * gst/rtsp-server/rtsp-auth.h:
8770 * gst/rtsp-server/rtsp-server.c:
8771 * gst/rtsp-server/rtsp-server.h:
8772 auth: move TLS handling to auth module
8773 Remove the TLS settings on the server and move it to the auth module because
8774 that is where security related bits go.
8776 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8778 * gst/rtsp-server/rtsp-client.c:
8779 * gst/rtsp-server/rtsp-client.h:
8780 client: add state push/pop
8782 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8784 * gst/rtsp-server/rtsp-client.c:
8785 * gst/rtsp-server/rtsp-client.h:
8786 client: add connection to state
8788 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8790 * gst/rtsp-server/rtsp-mount-points.c:
8791 mount-points: fix debug
8793 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8795 * tests/check/gst/media.c:
8796 tests: fix media test
8798 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8800 * gst/rtsp-server/rtsp-thread-pool.c:
8801 thread-pool: we don't require a state
8803 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8805 * gst/rtsp-server/rtsp-server.c:
8806 server: let context ref the server
8807 So that we don't risk losing the server object early anc crash.
8809 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8811 * tests/check/gst/client.c:
8812 tests: fix client test
8814 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8817 * docs/libs/gst-rtsp-server-docs.sgml:
8818 * docs/libs/gst-rtsp-server-sections.txt:
8819 * gst/rtsp-server/rtsp-address-pool.c:
8820 * gst/rtsp-server/rtsp-auth.c:
8821 * gst/rtsp-server/rtsp-client.c:
8822 * gst/rtsp-server/rtsp-client.h:
8823 * gst/rtsp-server/rtsp-media-factory-uri.c:
8824 * gst/rtsp-server/rtsp-media-factory.c:
8825 * gst/rtsp-server/rtsp-media-factory.h:
8826 * gst/rtsp-server/rtsp-media.c:
8827 * gst/rtsp-server/rtsp-mount-points.c:
8828 * gst/rtsp-server/rtsp-params.c:
8829 * gst/rtsp-server/rtsp-permissions.c:
8830 * gst/rtsp-server/rtsp-sdp.c:
8831 * gst/rtsp-server/rtsp-server.c:
8832 * gst/rtsp-server/rtsp-server.h:
8833 * gst/rtsp-server/rtsp-session-media.c:
8834 * gst/rtsp-server/rtsp-session-pool.c:
8835 * gst/rtsp-server/rtsp-session.c:
8836 * gst/rtsp-server/rtsp-stream-transport.c:
8837 * gst/rtsp-server/rtsp-stream.c:
8838 * gst/rtsp-server/rtsp-thread-pool.c:
8839 * gst/rtsp-server/rtsp-token.c:
8842 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-session-pool.c:
8845 * gst/rtsp-server/rtsp-session-pool.h:
8846 session-pool: make vmethod to create a session
8847 Make a vmethod to create a sessions so that subclasses can create
8848 custom session objects
8850 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8852 * gst/rtsp-server/rtsp-auth.c:
8853 * gst/rtsp-server/rtsp-media-factory.h:
8854 * gst/rtsp-server/rtsp-media.h:
8855 * gst/rtsp-server/rtsp-mount-points.h:
8856 * gst/rtsp-server/rtsp-session-pool.h:
8857 * gst/rtsp-server/rtsp-stream.h:
8860 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8862 * docs/libs/gst-rtsp-server-docs.sgml:
8863 * docs/libs/gst-rtsp-server-sections.txt:
8864 * gst/rtsp-server/rtsp-address-pool.c:
8865 * gst/rtsp-server/rtsp-address-pool.h:
8866 * gst/rtsp-server/rtsp-auth.c:
8867 * gst/rtsp-server/rtsp-client.h:
8868 * gst/rtsp-server/rtsp-media-factory.h:
8869 * gst/rtsp-server/rtsp-media.c:
8870 * gst/rtsp-server/rtsp-media.h:
8871 * gst/rtsp-server/rtsp-permissions.c:
8872 * gst/rtsp-server/rtsp-permissions.h:
8873 * gst/rtsp-server/rtsp-server.h:
8874 * gst/rtsp-server/rtsp-session-media.c:
8875 * gst/rtsp-server/rtsp-session-media.h:
8876 * gst/rtsp-server/rtsp-session-pool.h:
8877 * gst/rtsp-server/rtsp-session.h:
8878 * gst/rtsp-server/rtsp-stream-transport.h:
8879 * gst/rtsp-server/rtsp-stream.c:
8880 * gst/rtsp-server/rtsp-thread-pool.h:
8883 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8886 * examples/Makefile.am:
8887 configure: compile cgroup example conditionally
8888 Only compile the cgroup example when we have libcgroup
8890 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8893 * examples/Makefile.am:
8894 * examples/test-cgroups.c:
8895 examples: add cgroups example
8897 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8899 * tests/check/gst/rtspserver.c:
8900 tests: fix compilation
8902 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8904 * gst/rtsp-server/rtsp-thread-pool.c:
8905 thread-pool: fix vmethod invocation
8907 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8909 * gst/rtsp-server/rtsp-thread-pool.c:
8910 * gst/rtsp-server/rtsp-thread-pool.h:
8911 thread-pool: store thread type in thread
8913 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8915 * gst/rtsp-server/rtsp-client.c:
8916 client: pass thread from pool to media _prepare
8917 Get a thread from the configured threadpool and pass it to the prepare method of
8920 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8922 * gst/rtsp-server/rtsp-media.c:
8923 * gst/rtsp-server/rtsp-media.h:
8924 media: Accept a thread in _prepare
8925 Remove out own threadpool handling and use the provided thread and
8926 maincontext for the bus messages and the state changes.
8928 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8930 * gst/rtsp-server/rtsp-server.c:
8931 server: configure client thread pool
8933 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8935 * gst/rtsp-server/rtsp-client.c:
8936 * gst/rtsp-server/rtsp-client.h:
8937 client: add method to configure thread pool
8939 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8941 * gst/rtsp-server/rtsp-client.h:
8942 * gst/rtsp-server/rtsp-server.c:
8943 * gst/rtsp-server/rtsp-server.h:
8944 server: use thread pool
8945 Use the thread pool instead of doing our own thing.
8947 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8949 * gst/rtsp-server/Makefile.am:
8950 * gst/rtsp-server/rtsp-thread-pool.c:
8951 * gst/rtsp-server/rtsp-thread-pool.h:
8952 thread-pool: add object to manage threads
8953 Add an object to manage the client and media threads.
8955 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8957 * gst/rtsp-server/rtsp-auth.c:
8958 auth: debug authorization check
8960 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8962 * gst/rtsp-server/rtsp-media.c:
8963 media: start media pipeline in context
8964 Start the media pipeline in the provided context (or our default one
8965 when NULL). This makes sure that we run the bus thread in this context and that
8966 all media threads are children of this context.
8968 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * gst/rtsp-server/rtsp-media-factory.c:
8971 factory: pass permissions to media by default
8973 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8975 * examples/test-auth.c:
8976 test: add permissions to auth test
8977 Ass some permissions to the media factory in the test.
8979 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8981 * gst/rtsp-server/rtsp-auth.c:
8982 * gst/rtsp-server/rtsp-auth.h:
8983 * gst/rtsp-server/rtsp-client.c:
8984 auth: simplify auth checks
8985 Remove client from methods, it's now in the state
8986 Perform the check specified by the string, use the information from the
8987 thread local context.
8989 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * gst/rtsp-server/rtsp-client.c:
8992 * gst/rtsp-server/rtsp-client.h:
8993 client: add state to current thread
8994 Add the client to the ClientState object.
8995 Place the ClientState on the current thread.
8997 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8999 * gst/rtsp-server/rtsp-media-factory.c:
9000 * gst/rtsp-server/rtsp-media-factory.h:
9001 * gst/rtsp-server/rtsp-media.c:
9002 * gst/rtsp-server/rtsp-media.h:
9003 media: make it possible to set permissions
9004 Make it possible to set permissions on media and media factory objects
9006 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9008 * gst/rtsp-server/Makefile.am:
9009 * gst/rtsp-server/rtsp-permissions.c:
9010 * gst/rtsp-server/rtsp-permissions.h:
9011 permissions: add permissions object
9012 Add a mini object to store permissions based on a role.
9014 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9016 * examples/test-auth.c:
9017 * gst/rtsp-server/rtsp-auth.c:
9018 * gst/rtsp-server/rtsp-auth.h:
9019 * gst/rtsp-server/rtsp-client.c:
9020 auth: add auth checks
9021 Add an enum with auth checks and implement the checks in the auth object.
9022 Perform the checks from the client.
9024 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9026 * examples/test-auth.c:
9027 * gst/rtsp-server/rtsp-auth.c:
9028 * gst/rtsp-server/rtsp-auth.h:
9029 * gst/rtsp-server/rtsp-client.h:
9030 auth: use the token after authentication
9031 After we authenticated a user, keep the Token around in the state.
9033 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9035 * gst/rtsp-server/rtsp-client.c:
9036 * gst/rtsp-server/rtsp-media.c:
9037 * gst/rtsp-server/rtsp-media.h:
9038 * tests/check/gst/media.c:
9039 media: add optional context for bus messages
9040 Add an optional mainloop to _prepare that will handle the bus messages instead
9041 of always using the shared mainloop.
9043 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9045 * gst/rtsp-server/Makefile.am:
9046 * gst/rtsp-server/rtsp-token.c:
9047 * gst/rtsp-server/rtsp-token.h:
9048 token: add authorization token
9049 Add a simply miniobject that contains the authorizations. The object contains a
9050 GstStructure that hold all authorization fields. When a user is authenticated,
9051 the auth module will create a Token for the user. The token is then used to
9052 check what operations the user is allowed to do and various other configuration
9055 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9057 * examples/test-auth.c:
9058 * gst/rtsp-server/rtsp-auth.c:
9059 * gst/rtsp-server/rtsp-auth.h:
9060 * gst/rtsp-server/rtsp-client.c:
9061 * gst/rtsp-server/rtsp-client.h:
9062 * gst/rtsp-server/rtsp-media-factory.c:
9063 * gst/rtsp-server/rtsp-media-factory.h:
9064 * gst/rtsp-server/rtsp-media.c:
9065 * gst/rtsp-server/rtsp-media.h:
9066 auth: remove auth from media and factory
9067 Remove the auth object from media and factory. We want to have the RTSPClient
9068 authenticate and authorize resources, there is no need to place another auth
9069 manager on the media/factory.
9071 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9073 * examples/test-auth.c:
9074 * gst/rtsp-server/rtsp-auth.c:
9075 * gst/rtsp-server/rtsp-auth.h:
9076 * gst/rtsp-server/rtsp-client.h:
9077 auth: add support for multiple basic auth tokens
9078 Make it possible to add multiple basic authorisation tokens to one authorization
9079 object. Associate with each token an authorization group that will define what
9080 capabilities are allowed.
9082 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9084 * gst/rtsp-server/rtsp-client.c:
9085 client: error out on non-aggregate control
9086 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
9088 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9090 * gst/rtsp-server/rtsp-client.c:
9091 client: rework setup request a little
9092 Cache the media in DESCRIBE based on the longest matching path with the uri
9093 that we can find in the mount points.
9094 Rework the setup request a little to get the media from the session or from
9095 the longest matching path, this way we can derive the control string as
9096 everything after the path instead of hardcoding it.
9097 Find the stream based on the control string and only open a session when all
9100 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9102 * gst/rtsp-server/rtsp-media.c:
9103 * gst/rtsp-server/rtsp-media.h:
9104 media: add method to find a stream by control url
9106 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9108 * gst/rtsp-server/rtsp-stream.c:
9109 * gst/rtsp-server/rtsp-stream.h:
9110 stream: add method to check control url of stream
9112 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9114 * gst/rtsp-server/rtsp-client.c:
9115 * gst/rtsp-server/rtsp-session-media.c:
9116 * gst/rtsp-server/rtsp-session-media.h:
9117 * gst/rtsp-server/rtsp-session.c:
9118 * gst/rtsp-server/rtsp-session.h:
9119 session: use path matching for session media
9120 Use a path string instead of a uri to lookup session media in the sessions. Also
9121 use path matching to find the largest possible path that matches.
9123 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * gst/rtsp-server/rtsp-client.c:
9126 * gst/rtsp-server/rtsp-mount-points.c:
9127 * gst/rtsp-server/rtsp-mount-points.h:
9128 * tests/check/gst/mountpoints.c:
9129 mount-points: remove useless vmethod
9130 Making lookups in the mount points should not be done with a URL, if there is a
9131 mapping to be done from URL to mount points, we'll need to do it somewhere
9134 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9136 * gst/rtsp-server/rtsp-mount-points.c:
9137 * gst/rtsp-server/rtsp-mount-points.h:
9138 * tests/check/gst/mountpoints.c:
9139 mount-points: improve mount point searching
9140 Use a GSequence to keep track of the mount points.
9141 Match a URL to the longest matching registered mount point. This should be the
9142 URL to perform aggreagate control and the remainder is the stream specific
9144 Add some unit tests for this.
9146 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
9148 * gst/rtsp-server/Makefile.am:
9149 rtsp-server: Allow building of static library
9151 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9153 * tests/check/gst/mediafactory.c:
9154 tests: fix compilation
9156 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9158 * gst/rtsp-server/rtsp-sdp.c:
9159 sdp: get control string from stream
9160 Use the control string as configured in the stream.
9162 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9164 * gst/rtsp-server/rtsp-stream.c:
9165 * gst/rtsp-server/rtsp-stream.h:
9166 stream: add methods and property to set control string
9168 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9170 * gst/rtsp-server/rtsp-client.c:
9172 Rename variables for clarity
9173 Keep media in state when we can
9175 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * gst/rtsp-server/rtsp-client.c:
9178 * gst/rtsp-server/rtsp-stream.c:
9179 * gst/rtsp-server/rtsp-stream.h:
9180 stream: add more support for IPv6
9181 Rename _get_address to _get_multicast_address in GstRTSPStream to
9182 make it clear that this function only deals with multicast.
9183 Make it possible to have both an IPv4 and IPv6 multicast address on
9184 a stream. Give the client an IPv4 or IPv6 address depending on the
9185 address it used to connect to the server.
9186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
9188 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9190 * gst/rtsp-server/rtsp-client.c:
9193 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9195 * gst/rtsp-server/rtsp-stream.c:
9196 stream: handle failed port allocation
9197 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
9198 can't allocate any family at all. Also keep track of what port families we
9200 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
9202 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9204 * gst/rtsp-server/rtsp-stream.c:
9205 stream: improve docs
9207 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9209 * gst/rtsp-server/rtsp-stream-transport.c:
9210 stream-transport: remove old if 0 block
9212 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
9214 * tests/check/gst/client.c:
9216 gst_rtsp_client_get_uri() has been removed
9217 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
9219 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9221 * gst/rtsp-server/rtsp-client.c:
9222 * gst/rtsp-server/rtsp-client.h:
9223 client: add method to filter managed sessions
9224 Add a method to filter the sessions managed by this client connection.
9225 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
9227 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9229 * gst/rtsp-server/rtsp-client.c:
9230 * gst/rtsp-server/rtsp-client.h:
9231 client: remove _get_uri() method
9232 Remove the get_uri() method on the client. A client has no uri, the uri
9233 property is an internal property to manage the last cached media for
9236 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9238 * gst/rtsp-server/rtsp-media-factory.h:
9239 media-factory: fix typo
9241 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
9243 * gst/rtsp-server/rtsp-media.c:
9244 rtsp-media: Do not leak the query in default_query_stop
9245 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
9247 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9249 * gst/rtsp-server/rtsp-media.c:
9250 media: don't unlock when conversion fails
9251 Don't unlock the state lock when conversion fails because it was not locked.
9253 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9255 * gst/rtsp-server/rtsp-media.c:
9256 * gst/rtsp-server/rtsp-media.h:
9257 Add query_position and query_stop vmethods to rtsp-media
9259 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9261 * gst/rtsp-server/rtsp-media.c:
9262 Fix typo in property install for rtsp-media's time-provider
9264 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9266 * gst/rtsp-server/rtsp-client.c:
9267 * gst/rtsp-server/rtsp-client.h:
9268 client: clean some variables
9269 Clean some variables and add some guards to _send_request()
9271 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
9273 * gst/rtsp-server/rtsp-client.c:
9274 * gst/rtsp-server/rtsp-client.h:
9275 Add gst_rtsp_client_send_request API
9276 This makes it possible to send arbitrary messages to a client, such as
9277 SET_PARAMETER or GET_PARAMETER
9279 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9281 * gst/rtsp-server/rtsp-media.c:
9282 * gst/rtsp-server/rtsp-media.h:
9283 media: add _get_element() method
9284 Add method to get the element used when creating the media.
9285 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
9287 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9289 * gst/rtsp-server/rtsp-media.c:
9292 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9294 * gst/rtsp-server/rtsp-stream.c:
9295 * gst/rtsp-server/rtsp-stream.h:
9296 stream: allow access to the rtp session
9297 https://bugzilla.gnome.org/show_bug.cgi?id=703004
9299 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
9301 * gst/rtsp-server/rtsp-stream.c:
9302 * gst/rtsp-server/rtsp-stream.h:
9303 dscp qos support in gst-rtsp-stream
9304 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
9306 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * tests/check/gst/rtspserver.c:
9310 Actually do what the comment says. Also keep the old code around, not sure what
9311 should happen when you get a 454 from a TEARDOWN, does it close the connection?
9312 it currently doesn't.
9314 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9316 * gst/rtsp-server/rtsp-client.c:
9317 client: also watch newly created session
9318 When we newly created a session, start watching it immediately instead of
9319 on the next request.
9321 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
9323 * tests/check/gst/client.c:
9324 tests: add unit test for new-session
9325 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
9327 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9329 * gst/rtsp-server/rtsp-client.c:
9330 client: emit new-session when new session is created
9331 Only emit new-session when we created a new session for a client, not when a
9332 client picked up a previous session.
9333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
9335 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
9337 * gst/rtsp-server/rtsp-client.c:
9338 client: handle asterisk as path in requests
9339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
9341 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9343 * gst/rtsp-server/rtsp-media.c:
9344 media: handle segment query format mismatch
9345 It's possible that the segment query returns with a different format than what
9346 we asked for, handle this case also.
9348 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
9350 * gst/rtsp-server/rtsp-media.c:
9351 media: use segment stop in collect_media_stats
9352 Use segment stop instead of duration as range end point.
9353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
9355 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9357 * gst/rtsp-server/rtsp-media.c:
9358 * tests/check/gst/media.c:
9359 rtsp-media: Do not leak the element in take_pipeline
9360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
9362 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
9364 * gst/rtsp-server/rtsp-client.c:
9365 * gst/rtsp-server/rtsp-client.h:
9366 rtsp-client: Make configure_client_transport virtual
9367 This patch makes configure_client_transport virtual. The functionality is
9368 needed to handle some weird clients sending multicast transport settings as url
9370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
9372 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
9374 * gst/rtsp-server/rtsp-client.c:
9375 * gst/rtsp-server/rtsp-client.h:
9376 rtsp-client: Make param_set and param_get virtual
9377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
9379 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
9381 * gst/rtsp-server/rtsp-client.c:
9382 * gst/rtsp-server/rtsp-media.c:
9383 * gst/rtsp-server/rtsp-media.h:
9384 media: convert_range replaces get_range_times
9385 get_range_times worked for handling UTC ranges for seeks, but we also
9386 need to convert back from NPT to the requested unit in
9387 get_range_string. convert_range is now used for both.
9388 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
9390 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9392 * gst/rtsp-server/rtsp-client.c:
9393 * gst/rtsp-server/rtsp-sdp.c:
9394 * gst/rtsp-server/rtsp-sdp.h:
9395 sdp: cleanup sdp info
9396 We don't need to pass the proto, we can more easily check a boolean.
9397 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
9399 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
9401 * gst/rtsp-server/rtsp-sdp.c:
9402 use 0.0.0.0 or :: for c= line instead of server address
9404 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
9406 * gst/rtsp-server/rtsp-client.c:
9407 use local address, not remote, in SDP
9408 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
9410 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9413 Automatic update of common submodule
9414 From 098c0d7 to 01a7a46
9416 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
9418 * gst/rtsp-server/rtsp-media.c:
9419 * gst/rtsp-server/rtsp-media.h:
9420 media: possibility to override range time conversion
9421 Make it possible to override the conversion from GstRTSPTimeRange to
9422 GstClockTimes, that is done before seeking on the media
9423 pipeline. Overriding can be useful for UTC ranges, where the default
9424 conversion gives nanoseconds since 1900.
9425 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
9427 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
9429 * gst/rtsp-server/rtsp-server.c:
9430 * gst/rtsp-server/rtsp-server.h:
9431 rtsp-server: Expose the use_client_settings API
9432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
9434 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
9436 * gst/rtsp-server/rtsp-client.c:
9437 * gst/rtsp-server/rtsp-stream.c:
9438 * gst/rtsp-server/rtsp-stream.h:
9439 rtspstream: handle both ipv4 and ipv6 clients
9440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
9442 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9444 * gst/rtsp-server/rtsp-sdp.c:
9445 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
9446 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
9447 We already have a way to place extra attributes in the SDP by using a string
9448 property with prefix x- or a- in the caps.
9450 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9452 * gst/rtsp-server/rtsp-sdp.c:
9453 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
9454 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
9455 We already have a way to place extra attributes in the SDP, just make a string
9456 property in the payloader with a- or x- prefix.
9458 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9460 * gst/rtsp-server/rtsp-sdp.c:
9461 rtsp: place a- and x- properties as attributes
9462 application/x-rtp has properties with a- and x- prefixes that should be
9463 placed as attributes in the SDP for the media instead of being added to the
9466 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9468 * examples/Makefile.am:
9469 * examples/test-video.c:
9470 example: add TLS example
9472 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9474 * gst/rtsp-server/rtsp-server.c:
9475 * gst/rtsp-server/rtsp-server.h:
9476 server: add support for TLS
9477 Add methods to set and get a TLS certificate.
9478 Add vmethod to configure a new connection. By default, configure the TLS
9479 certificate in a new connection if needed.
9481 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9483 * gst/rtsp-server/rtsp-server.c:
9484 * gst/rtsp-server/rtsp-server.h:
9485 server: remove accept_client vmethod
9486 This vmethod is not very useful so remove it.
9488 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9490 * gst/rtsp-server/rtsp-server.c:
9491 server: don't crash on NULL GError
9493 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
9495 * gst/rtsp-server/rtsp-session-pool.c:
9496 rtsp-session-pool: corrected session timeout detection
9497 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
9499 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9501 * gst/rtsp-server/rtsp-client.c:
9502 client: improve debug
9504 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9506 * gst/rtsp-server/rtsp-client.c:
9507 * gst/rtsp-server/rtsp-client.h:
9508 * gst/rtsp-server/rtsp-server.c:
9509 server: refactor connection setup
9510 Let the server accept the socket connection and construct a GstRTSPConnection
9511 from it. Remove the code from the client and let the client only deal with
9512 a fully configure GstRTSPConnection object.
9513 We will need this later when the server will configure the connection for
9516 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9518 * gst/rtsp-server/rtsp-stream.c:
9519 stream: keep the transport object alive
9520 Keep the transport object alive while we have it as qdata on the
9523 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
9525 * gst/rtsp-server/rtsp-client.c:
9526 * gst/rtsp-server/rtsp-server.c:
9527 rtsp-server: Do not crash on nmapping of server
9528 * generate error when gst_rtsp_connection_accept fails
9529 * do not stop accepting incoming connections because
9530 accepting a client fails
9531 https://bugzilla.gnome.org/show_bug.cgi?id=701072
9533 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
9535 * gst/rtsp-server/rtsp-client.c:
9536 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
9537 https://bugzilla.gnome.org/show_bug.cgi?id=700953
9539 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9541 * gst/rtsp-server/rtsp-sdp.c:
9542 rtsp-sdp: Parse framerate caps field and set SDP attribute
9543 The SDP attribute and its format is described in RFC4566.
9544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9546 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
9548 * gst/rtsp-server/rtsp-sdp.c:
9549 rtsp-sdp: Parse width/height from caps and set SDP attribute
9550 The SDP attribute and its format is described in RFC6064.
9551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
9553 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
9555 * gst/rtsp-server/rtsp-sdp.c:
9556 * tests/check/gst/client.c:
9557 rtsp-sdp: add bandwidth line
9558 https://bugzilla.gnome.org/show_bug.cgi?id=699220
9560 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9563 Automatic update of common submodule
9564 From 5edcd85 to 098c0d7
9566 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9568 * tests/check/gst/media.c:
9569 tests: add dynamic payloader prepare/unprepare check
9571 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9573 * gst/rtsp-server/rtsp-media.c:
9574 media: release lock when removing fakesink
9576 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9578 * gst/rtsp-server/rtsp-stream.c:
9579 stream: set elements to NULL before removing
9580 When removing a stream, set the elements to NULL first. This avoids
9581 element-is-not-in-NULL-state errors when we dispose the elements.
9583 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9586 Automatic update of common submodule
9587 From 3cb3d3c to 5edcd85
9589 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9591 * gst/rtsp-server/rtsp-media.c:
9592 * gst/rtsp-server/rtsp-media.h:
9593 media: listen to pad-removed signals
9594 Listen to the pad-removed signal and remove the stream associated with the
9596 Add signal to be notified of the removed pad.
9597 Remove the fakesink in unprepare()
9598 Fix signatures of the signal methods
9600 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9602 * examples/test-sdp.c:
9603 tests: add example of reusable pipelines
9605 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
9607 * gst/rtsp-server/rtsp-stream.c:
9608 * gst/rtsp-server/rtsp-stream.h:
9609 stream: add method to get the srcpad
9611 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
9613 * tests/check/gst/media.c:
9614 check: add media prepare/unprepare test
9615 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9617 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
9619 * gst/rtsp-server/rtsp-media.c:
9620 media: disconnect from signal handlers in unprepare()
9621 We connected to the pad-added and no-more-pads signals in prepare() so
9622 we need to disconnect from them in unprepare().
9623 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9625 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
9627 * gst/rtsp-server/rtsp-media.c:
9628 media: don't free streams array
9629 Don't free the streams array in the unprepare() method, they were not
9631 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9633 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
9635 * gst/rtsp-server/rtsp-media.c:
9636 media: don't unref the pipeline in unprepare
9637 Unprepare() should undo what prepare() does. Because the pipeline is
9638 not created in prepare(), we should not unref it in unprepare()
9640 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
9642 * gst/rtsp-server/rtsp-stream.c:
9643 stream: clear session and caps for reuse
9644 Set the session and caps to NULL after unref otherwise we might unref
9646 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
9648 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
9650 * gst/rtsp-server/rtsp-client.c:
9651 client: send out teardown signal before tearing down
9652 The advantage is that in the signal handler you get direct access to
9653 information about what streams are about to get torn down (in the
9654 GstRTSPClientState).
9655 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
9657 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
9659 * gst/rtsp-server/rtsp-client.c:
9660 * gst/rtsp-server/rtsp-client.h:
9661 client: expose connection
9662 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
9664 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
9667 Automatic update of common submodule
9668 From aed87ae to 3cb3d3c
9670 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9672 * gst/rtsp-server/rtsp-media.c:
9673 * gst/rtsp-server/rtsp-media.h:
9674 * gst/rtsp-server/rtsp-session-media.c:
9675 * gst/rtsp-server/rtsp-session-media.h:
9676 media: add method to get the base_time of the pipeline
9677 Together with a shared clock, this base-time could eventually be sent to
9678 the client so that it can reconstruct the exact running-time of the clock
9681 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9683 * gst/rtsp-server/Makefile.am:
9684 * gst/rtsp-server/rtsp-media.c:
9685 * gst/rtsp-server/rtsp-media.h:
9686 * gst/rtsp-server/rtsp-sdp.c:
9687 media: add GstNetTimeProvider support
9688 Add a property to let the media provide a GstNetTimeProvider for its clock.
9689 Make methods to get the clock and nettimeprovider
9690 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
9691 provider and also the current time of the clock. This should make it possible
9692 for (GStreamer) clients to slave their clock to the server clock.
9694 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
9697 Automatic update of common submodule
9698 From 04c7a1e to aed87ae
9700 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9702 * gst/rtsp-server/rtsp-media.c:
9703 media: wait for buffering to complete
9704 Wait for buffering to complete before changing the state to the target state.
9706 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9708 * gst/rtsp-server/rtsp-media.c:
9709 media: small cleanup
9711 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
9713 * tests/check/gst/rtspserver.c:
9714 tests: remove extra unref in test_setup_non_existing_stream
9715 The unref is not needed anymore, teardown runs without it.
9716 https://bugzilla.gnome.org/show_bug.cgi?id=696542
9718 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
9720 * tests/check/gst/rtspserver.c:
9721 tests: GSocketService cleanup in test_bind_already_in_use
9722 Use g_socket_service_stop so the rtspserver test stops listening for
9723 incoming connections in test_bind_already_in_use.
9724 https://bugzilla.gnome.org/show_bug.cgi?id=696541
9726 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
9728 * gst/rtsp-server/rtsp-media-factory.c:
9729 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
9730 Instead use a GWeakRef which is safe to use
9731 This is a known GLib bug, see:
9732 https://bugzilla.gnome.org/show_bug.cgi?id=667145
9734 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
9736 * gst/rtsp-server/rtsp-client.c:
9737 * gst/rtsp-server/rtsp-media.c:
9738 * gst/rtsp-server/rtsp-media.h:
9739 * gst/rtsp-server/rtsp-sdp.c:
9740 * tests/check/gst/media.c:
9741 * tests/check/gst/rtspserver.c:
9742 rtsp-media/client: Reply to PLAY request with same type of Range
9743 Remember the type of Range from the PLAY request and use the same type for
9746 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
9748 * gst/rtsp-server/rtsp-client.c:
9749 * gst/rtsp-server/rtsp-client.h:
9750 * tests/check/gst/client.c:
9751 rtsp-client: expose uri
9753 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
9755 * tests/check/gst/mediafactory.c:
9756 tests: Hold ref while creating second media
9757 To test if the media aren't shared, make sure we keep the first one while creating a second
9758 otherwise the same memory address may be reused.
9760 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
9763 configure: remove out-of-date comment
9765 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
9768 .gitignore: ignore more build files
9770 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
9772 * tests/check/Makefile.am:
9773 tests: use right _LIBS variable for gst-plugins-base libs
9775 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9777 * tests/check/Makefile.am:
9778 check: add librtp to libs
9780 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
9782 * tests/check/gst/rtspserver.c:
9783 tests: Add test to check selecting a port the server will send from
9785 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
9787 * tests/check/gst/rtspserver.c:
9788 tests: Make sure packets are actually received
9790 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9792 * gst/rtsp-server/rtsp-stream.c:
9793 stream: Select unicast address from pool if appropriate
9795 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
9797 * gst/rtsp-server/rtsp-stream.c:
9798 stream: Properties are always there in Gst 1.0
9800 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9802 * tests/check/gst/addresspool.c:
9803 tests: Add tests for unicast addresses in pool
9805 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
9807 * gst/rtsp-server/rtsp-address-pool.c:
9808 * tests/check/gst/addresspool.c:
9809 address-pool: Verify that multicast addresses are used for multicast and vice-versa
9811 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
9813 * docs/libs/gst-rtsp-server-sections.txt:
9814 * gst/rtsp-server/rtsp-address-pool.c:
9815 * gst/rtsp-server/rtsp-address-pool.h:
9816 * gst/rtsp-server/rtsp-stream.c:
9817 * tests/check/gst/addresspool.c:
9818 address-pool: Add unicast addresses
9820 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9823 * gst/rtsp-server/rtsp-server.c:
9824 * tests/check/gst/rtspserver.c:
9825 rtsp-server: Limit the number of threads per server instance
9826 If we exceed the maximum, just round robin the clients over the existing
9829 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
9831 * gst/rtsp-server/rtsp-server.c:
9832 rtsp-server: No need to store the GMainContext in the client context
9834 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
9836 * tests/check/gst/rtspserver.c:
9837 tests: Add test for client disconnection
9839 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
9841 * tests/check/gst/rtspserver.c:
9842 tests: Test client and session timeouts with multiple threads
9844 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
9846 * gst/rtsp-server/rtsp-address-pool.c:
9847 * gst/rtsp-server/rtsp-auth.c:
9848 * gst/rtsp-server/rtsp-client.c:
9849 * gst/rtsp-server/rtsp-media-factory-uri.c:
9850 * gst/rtsp-server/rtsp-media-factory.c:
9851 * gst/rtsp-server/rtsp-media.c:
9852 * gst/rtsp-server/rtsp-mount-points.c:
9853 * gst/rtsp-server/rtsp-server.c:
9854 * gst/rtsp-server/rtsp-session-media.c:
9855 * gst/rtsp-server/rtsp-session-pool.c:
9856 * gst/rtsp-server/rtsp-session.c:
9857 Document locking and its order
9859 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
9861 * tests/check/gst/rtspserver.c:
9862 tests: Test that slow DESCRIBE don't block other clients
9864 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
9866 * tests/check/gst/client.c:
9867 tests: Add tests for client-requested multicast address
9869 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
9871 * docs/libs/gst-rtsp-server-sections.txt:
9872 docs: Put the various functions in the right sections
9874 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
9876 * docs/libs/gst-rtsp-server-docs.sgml:
9877 * docs/libs/gst-rtsp-server-sections.txt:
9878 * gst/rtsp-server/rtsp-address-pool.c:
9879 * gst/rtsp-server/rtsp-address-pool.h:
9880 docs: Generate docs for GstRTSPAddressPool
9882 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
9884 * gst/rtsp-server/rtsp-client.c:
9885 * gst/rtsp-server/rtsp-stream.c:
9886 * gst/rtsp-server/rtsp-stream.h:
9887 client: Check client provided addresses against the address pool
9889 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
9891 * gst/rtsp-server/rtsp-address-pool.c:
9892 * gst/rtsp-server/rtsp-address-pool.h:
9893 * tests/check/gst/addresspool.c:
9894 address-pool: Add API to request a specific address from the pool
9895 Also add relevant unit tests.
9897 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
9899 * tests/check/gst/mediafactory.c:
9900 tests: Check the passing around of a RTSPAddressPool
9901 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
9902 way down to the stream.
9904 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
9906 * tests/check/gst/addresspool.c:
9907 tests: Add more tests for the address pool
9909 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
9911 * gst/rtsp-server/rtsp-address-pool.c:
9912 address-pool: Fix off by one error
9913 When splitting a port range, the port after a skip is not part of range.
9915 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
9918 Automatic update of common submodule
9919 From 2de221c to 04c7a1e
9921 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
9924 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
9925 AM_CONFIG_HEADER was removed in automake 1.13
9926 https://bugzilla.gnome.org/show_bug.cgi?id=693368
9928 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
9931 Automatic update of common submodule
9932 From a942293 to 2de221c
9934 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9936 * gst/rtsp-server/rtsp-client.c:
9937 client: make sure the watch exists while sending data
9938 Protect the send_func with a lock. This allows us to wait for sending
9939 to complete before changing the send_func and user_data. We add an
9940 extra ref to the watch to make sure that it remains valid during
9942 When closing the connection, set the send_func to NULL
9943 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
9945 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9947 * tests/check/Makefile.am:
9948 tests: use GST_*_1_0 environment variables everywhere
9949 The _1_0 suffixed environment variables override the
9950 non-suffixed ones, so if we're in an environment that
9951 sets the _1_0 suffixed ones, such as jhbuild, we need
9952 to set those to make sure ours actually always get
9955 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9958 Automatic update of common submodule
9959 From acb04d9 to a942293
9961 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9963 * gst/rtsp-server/rtsp-client.c:
9964 rtsp-client: set the client backlog
9965 Set the client backlog to a reasonable default
9967 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
9969 * gst/rtsp-server/rtsp-media.c:
9970 rtsp-media: Make the element a constructor parameter
9971 https://bugzilla.gnome.org/show_bug.cgi?id=689594
9973 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
9975 * docs/libs/Makefile.am:
9976 docs: Link with gcov library when gcov is enabled
9977 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
9979 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9981 * gst/rtsp-server/rtsp-media.c:
9982 media: match prepare with unprepare
9983 Really unprepare when there were an equal amount of prepare calls.
9985 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9987 * gst/rtsp-server/rtsp-media.c:
9988 media: media has to be unprepared in finalize
9989 Because unprepare takes away the last ref on the media.
9991 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9993 * gst/rtsp-server/rtsp-client.c:
9994 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
9995 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
9996 We can't use the refcount to trigger unprepare because it is the unprepare call
9997 that removes the last refcount after all messages are consumed. What we should
9998 probably do is make a prepared refcount and only unprepare when the refcount
10001 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10003 * gst/rtsp-server/rtsp-media.c:
10004 media: let the source unref the last media ref
10005 the last ref to the media is held by the source so we don't need to add more ref
10006 and unrefs, we simply destroy the media when the source is gone.
10008 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10010 * gst/rtsp-server/rtsp-media.c:
10011 media: improve debug
10013 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10015 * gst/rtsp-server/rtsp-media.c:
10017 Make sure we are in the right state when collecting the position and duration.
10018 Only make ourselves PREPARED when we were previously PREPARING.
10020 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10022 * gst/rtsp-server/rtsp-media.c:
10023 media: use g_object_ref/unref for GObjects
10025 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
10027 * gst/rtsp-server/rtsp-client.c:
10028 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
10029 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
10030 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
10031 isn't being used anymore.
10033 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
10035 * gst/rtsp-server/rtsp-media.c:
10036 Fix compiler warning
10038 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
10040 * gst/rtsp-server/rtsp-media-factory-uri.c:
10041 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
10043 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10045 * gst/rtsp-server/rtsp-session-media.h:
10048 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10050 * gst/rtsp-server/rtsp-media.c:
10051 * tests/check/gst/media.c:
10052 media: avoid element leak
10054 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10056 * gst/rtsp-server/rtsp-media.c:
10057 media: require an element in media constructor
10059 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10061 * gst/rtsp-server/rtsp-client.c:
10062 Revert "client: TEARDOWN brings that state to Init again"
10063 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
10064 The object is already disposed, there is no point in setting the state.
10066 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10068 * gst/rtsp-server/rtsp-client.c:
10069 client: TEARDOWN brings that state to Init again
10071 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10073 * docs/libs/gst-rtsp-server-sections.txt:
10074 * examples/test-auth.c:
10075 * gst/rtsp-server/rtsp-auth.c:
10076 * gst/rtsp-server/rtsp-auth.h:
10077 * gst/rtsp-server/rtsp-client.c:
10078 * gst/rtsp-server/rtsp-client.h:
10079 * gst/rtsp-server/rtsp-media-factory-uri.c:
10080 * gst/rtsp-server/rtsp-media-factory-uri.h:
10081 * gst/rtsp-server/rtsp-media-factory.c:
10082 * gst/rtsp-server/rtsp-media-factory.h:
10083 * gst/rtsp-server/rtsp-media.c:
10084 * gst/rtsp-server/rtsp-media.h:
10085 * gst/rtsp-server/rtsp-mount-points.c:
10086 * gst/rtsp-server/rtsp-mount-points.h:
10087 * gst/rtsp-server/rtsp-sdp.c:
10088 * gst/rtsp-server/rtsp-server.c:
10089 * gst/rtsp-server/rtsp-server.h:
10090 * gst/rtsp-server/rtsp-session-media.c:
10091 * gst/rtsp-server/rtsp-session-media.h:
10092 * gst/rtsp-server/rtsp-session-pool.c:
10093 * gst/rtsp-server/rtsp-session-pool.h:
10094 * gst/rtsp-server/rtsp-session.c:
10095 * gst/rtsp-server/rtsp-session.h:
10096 * gst/rtsp-server/rtsp-stream-transport.c:
10097 * gst/rtsp-server/rtsp-stream-transport.h:
10098 * gst/rtsp-server/rtsp-stream.c:
10099 * gst/rtsp-server/rtsp-stream.h:
10100 * tests/check/gst/media.c:
10101 rtsp: make object details private
10102 Make all object details private
10103 Add methods to access private bits
10105 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10107 * tests/check/Makefile.am:
10108 * tests/check/gst/media.c:
10109 tests: add media tests
10111 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10113 * gst/rtsp-server/rtsp-media.c:
10114 media: check if prepared for some methods
10115 Check that the media object is prepared before doing seek and getting the
10116 current position etc.
10117 Add some g_return checks.
10119 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10121 * tests/check/Makefile.am:
10122 * tests/check/gst/mediafactory.c:
10123 tests: add mediafactory test
10125 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10127 * gst/rtsp-server/rtsp-stream.c:
10128 stream: improve debug
10130 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10132 * gst/rtsp-server/rtsp-media.c:
10133 * gst/rtsp-server/rtsp-media.h:
10134 media: unref pipeline in finalize to avoid leaking it
10136 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10138 * gst/rtsp-server/rtsp-media-factory-uri.c:
10139 * gst/rtsp-server/rtsp-media.c:
10140 rtsp: use gst_object_unref on GstObjects
10142 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10144 * gst/rtsp-server/rtsp-media-factory.c:
10145 media-factory: require an url
10147 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10149 * examples/test-uri.c:
10150 examples: fix include
10152 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10154 * gst/rtsp-server/rtsp-server.h:
10155 server: remove unused include
10157 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10159 * tests/check/Makefile.am:
10160 * tests/check/gst/mountpoints.c:
10161 tests: add test for mountpoints
10163 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10165 * gst/rtsp-server/rtsp-client.c:
10166 client: fix factory leak
10167 Keep the factory in the state object only for authorization checks and make
10168 sure we unref it on failure. Also don't keep invalid objects in the state
10171 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10173 * gst/rtsp-server/rtsp-mount-points.c:
10174 mounts: add g_return_if guards
10176 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10178 * tests/check/gst/client.c:
10179 tests: add more tests
10181 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10183 * gst/rtsp-server/rtsp-client.c:
10184 client: improve debug
10186 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10188 * gst/rtsp-server/rtsp-client.c:
10189 client: improve debug and fix leaks
10190 Cleanup the uri and session when there is a bad request.
10192 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10197 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10199 * tests/check/gst/client.c:
10200 test: add test for session in options request
10202 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10204 * gst/rtsp-server/rtsp-client.c:
10205 client: use 454 when session can't be found
10206 We should use 454 when a session can't be found because there was no session
10207 pool configured in the server. This is not a server configuration problem
10208 because the server on which the request is done might not be the same one that
10209 will keep the sessions for us and so it does not need to support sessions.
10211 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10213 * gst/rtsp-server/rtsp-client.c:
10214 client: only free connection when there is one
10215 It's possible that the client doesn't have a connection when we try to free it.
10217 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10219 * tests/check/Makefile.am:
10220 * tests/check/gst/client.c:
10221 tests: add unit test for the client object
10223 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10225 * gst/rtsp-server/rtsp-client.c:
10226 client: small cleanup
10228 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10230 * gst/rtsp-server/rtsp-client.h:
10231 client: remove unused include
10233 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10235 * gst/rtsp-server/rtsp-client.c:
10236 client: fix compilation
10238 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10240 * gst/rtsp-server/rtsp-client.c:
10241 client: call destroy without the lock
10243 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10245 * gst/rtsp-server/rtsp-client.c:
10246 * gst/rtsp-server/rtsp-client.h:
10247 client: make the client usable without a socket
10248 Make a method to let the client handle a message and a callback when the client
10249 wants us to send a response message back. This makes it possible to also use the
10250 client object without the sockets, which should make it easier to test.
10252 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10254 * gst/rtsp-server/rtsp-client.c:
10255 * gst/rtsp-server/rtsp-client.h:
10256 client: small cleanup
10258 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10260 * docs/libs/gst-rtsp-server-sections.txt:
10261 * gst/rtsp-server/rtsp-client.c:
10262 * gst/rtsp-server/rtsp-client.h:
10263 * gst/rtsp-server/rtsp-server.c:
10264 client: remove reference to server
10265 We don't need to keep a ref to the server
10267 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10269 * gst/rtsp-server/rtsp-client.c:
10270 * gst/rtsp-server/rtsp-client.h:
10271 client: add locking
10272 Also add some g_return_if()
10274 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10276 * gst/rtsp-server/rtsp-client.c:
10277 client: log more errors
10279 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10281 * gst/rtsp-server/rtsp-client.c:
10282 client: fix compilation
10284 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10286 * gst/rtsp-server/rtsp-client.c:
10287 * gst/rtsp-server/rtsp-client.h:
10288 client: add generic close-after-send support
10289 Add a property to send_response() to close the connection after the response has
10290 been sent to the client.
10292 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10295 * docs/libs/gst-rtsp-server-docs.sgml:
10296 * docs/libs/gst-rtsp-server-sections.txt:
10297 * docs/libs/gst-rtsp-server.types:
10298 * examples/test-auth.c:
10299 * examples/test-launch.c:
10300 * examples/test-mp4.c:
10301 * examples/test-multicast.c:
10302 * examples/test-multicast2.c:
10303 * examples/test-ogg.c:
10304 * examples/test-readme.c:
10305 * examples/test-sdp.c:
10306 * examples/test-uri.c:
10307 * examples/test-video.c:
10308 * gst/rtsp-server/Makefile.am:
10309 * gst/rtsp-server/rtsp-auth.h:
10310 * gst/rtsp-server/rtsp-client.c:
10311 * gst/rtsp-server/rtsp-client.h:
10312 * gst/rtsp-server/rtsp-media-mapping.c:
10313 * gst/rtsp-server/rtsp-media-mapping.h:
10314 * gst/rtsp-server/rtsp-mount-points.c:
10315 * gst/rtsp-server/rtsp-mount-points.h:
10316 * gst/rtsp-server/rtsp-server.c:
10317 * gst/rtsp-server/rtsp-server.h:
10318 * gst/rtsp-server/rtsp-session-media.c:
10319 * gst/rtsp-server/rtsp-session-pool.c:
10320 * gst/rtsp-server/rtsp-session-pool.h:
10321 * tests/check/gst/rtspserver.c:
10322 MediaMapping -> MountPoints
10323 Describes better what the object manages.
10325 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10328 configure: bump required version of -base
10330 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10332 * gst/rtsp-server/rtsp-media.c:
10335 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10337 * gst/rtsp-server/rtsp-media.c:
10338 * gst/rtsp-server/rtsp-media.h:
10339 media: support more Range formats
10340 Use the new -base methods to convert the Range string into a seek start and stop
10343 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10345 * examples/test-launch.c:
10346 examples: fix whitespace
10348 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10350 * examples/test-auth.c:
10351 test-auth: add example of how to remove sessions
10352 Add an example of the session filter api.
10354 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10356 * examples/test-uri.c:
10357 test-uri: remove mapping example
10359 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10361 * examples/test-uri.c:
10362 test-uri: fix callback signature
10364 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10366 * gst/rtsp-server/rtsp-media-factory.c:
10367 factory: keep ref to factory while media active
10368 While the media from a factory is alive, keep a ref to the factory.
10369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
10371 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10373 * gst/rtsp-server/rtsp-media-factory-uri.c:
10374 factory-uri: add some debug
10376 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10378 * gst/rtsp-server/rtsp-stream.c:
10379 stream: set udp sources to PLAYING
10380 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
10381 so that it doesn't cause our pipeline to produce ASYNC-DONE.
10383 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10385 * gst/rtsp-server/rtsp-media-factory-uri.c:
10386 factory-uri: take ref to factory
10387 Take a ref to the factory that we place in our list.
10389 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10391 * tests/Makefile.am:
10392 * tests/test-reuse.c:
10393 test: add test for server reuse
10394 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
10396 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
10398 * gst/rtsp-server/rtsp-server.c:
10399 server: start and stop multiple times
10400 Stop listening on the RTSP port when the GSource is removed, so clients
10401 can't connect and the server can be started again.
10402 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
10404 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10406 * gst/rtsp-server/rtsp-server.c:
10407 server: fix small leak
10409 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10411 * gst/rtsp-server/rtsp-media.c:
10412 media: unref source in finish_unprepare
10413 The source is created in prepare, unref it in finish_unprepare.
10414 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
10416 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
10418 * gst/rtsp-server/rtsp-client.c:
10419 * gst/rtsp-server/rtsp-media.c:
10420 rtsp-media: remove bus watch before finalizing
10421 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
10422 * An extra media ref is added for the bus watch. This extra ref is unreffed by
10423 the GDestroyNotify function.
10424 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
10425 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
10426 gst_rtsp_media_unprepare before unreffing the media.
10427 This way, the bus watch will be removed before the media is finalized.
10428 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
10430 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
10432 * gst/rtsp-server/rtsp-client.c:
10433 * gst/rtsp-server/rtsp-client.h:
10434 client: wait until the TEARDOWN response is sent to close the connection
10435 Responses can be sent async so we need to wait until the TEARDOWN response has
10436 been written before we close the connection to the client. This avoids the risk
10437 of writing/polling closed sockets.
10438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
10440 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
10442 * gst/rtsp-server/rtsp-stream.c:
10443 rtsp-stream: plug socket leak
10444 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
10446 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
10449 Automatic update of common submodule
10450 From 6bb6951 to a72faea
10452 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
10454 * gst/rtsp-server/rtsp-media-factory-uri.c:
10455 rtsp-server: don't use deprecated API
10457 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10459 * gst/rtsp-server/rtsp-client.c:
10460 rtsp-client: fix unused-but-set-variable compiler warning
10461 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
10463 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10466 * docs/libs/gst-rtsp-server-sections.txt:
10467 * gst/rtsp-server/rtsp-client.c:
10470 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10472 * examples/Makefile.am:
10473 * examples/test-multicast2.c:
10474 examples: add another multicast example
10475 Add an example for how to configure separate multicast ranges for each media
10478 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10480 * examples/test-multicast.c:
10483 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10485 * gst/rtsp-server/rtsp-client.c:
10486 * gst/rtsp-server/rtsp-media.c:
10487 * gst/rtsp-server/rtsp-session-media.c:
10488 * gst/rtsp-server/rtsp-session-media.h:
10489 * gst/rtsp-server/rtsp-stream-transport.c:
10490 * gst/rtsp-server/rtsp-stream-transport.h:
10491 stream: use the address managed by the stream
10492 Use the address managed by the stream for multicast. This allows us to have 1
10493 multicast address for each stream.
10494 Because the address is now managed by the stream we don't have to pass it around
10496 Set the address pool on the streams.
10498 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10500 * gst/rtsp-server/rtsp-client.c:
10501 * gst/rtsp-server/rtsp-media.c:
10502 * gst/rtsp-server/rtsp-stream.c:
10503 rtsp: improve debug
10505 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10507 * gst/rtsp-server/rtsp-media.c:
10508 * gst/rtsp-server/rtsp-media.h:
10509 media: add signal for new streams
10510 This allows applications to listen for new streams and configure properties on
10511 them, like the address pool.
10513 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10515 * gst/rtsp-server/rtsp-media.c:
10516 media: configure address pool in new streams
10518 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10520 * gst/rtsp-server/rtsp-stream.c:
10521 * gst/rtsp-server/rtsp-stream.h:
10522 stream: add methods to deal with address pool
10523 Add methods to get and set the address pool for the stream
10524 Add method to allocate and get the multicast addresses for this stream.
10526 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10528 * docs/libs/gst-rtsp-server-sections.txt:
10529 * gst/rtsp-server/rtsp-media.c:
10530 * gst/rtsp-server/rtsp-media.h:
10531 media: remove MTU property
10532 It is a stream property
10534 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10536 * gst/rtsp-server/rtsp-client.c:
10537 client: set blocksize only on stream
10538 Set the blocksize only on the current stream.
10540 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10542 * gst/rtsp-server/rtsp-stream.c:
10543 stream: share src and sink sockets
10544 the allocated socket is in the used-socket property, not socket.
10546 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10548 * gst/rtsp-server/rtsp-address-pool.c:
10549 * gst/rtsp-server/rtsp-address-pool.h:
10550 * gst/rtsp-server/rtsp-client.c:
10551 * gst/rtsp-server/rtsp-session-media.c:
10552 * gst/rtsp-server/rtsp-session-media.h:
10553 * gst/rtsp-server/rtsp-stream-transport.c:
10554 * gst/rtsp-server/rtsp-stream-transport.h:
10555 * tests/check/gst/addresspool.c:
10556 rtsp: make address-pool return an address object
10557 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
10558 store more info in the structure and allows us to more easily return the address
10559 to the right pool when no longer needed.
10560 Pass the address to the StreamTransport so that we can return it to the pool
10561 when the stream transport is freed or changed.
10563 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10565 * examples/Makefile.am:
10566 * examples/test-multicast.c:
10567 examples: add multicast example
10568 Show how to set up the multicast address pool so that media can be
10569 server with multicast.
10571 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10573 * gst/rtsp-server/rtsp-client.c:
10574 * gst/rtsp-server/rtsp-media-factory.c:
10575 * gst/rtsp-server/rtsp-media-factory.h:
10576 * gst/rtsp-server/rtsp-media.c:
10577 * gst/rtsp-server/rtsp-media.h:
10578 rtsp: use AddressPool
10579 Remove the multicast_group property.
10580 Use the configured addresspool to allocate multicast addresses.
10582 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10584 * gst/rtsp-server/rtsp-address-pool.c:
10585 * gst/rtsp-server/rtsp-address-pool.h:
10586 address-pool: add clear method
10588 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10590 * gst/rtsp-server/rtsp-address-pool.c:
10591 address-pool: small cleanups
10593 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10595 * tests/check/Makefile.am:
10596 * tests/check/gst/addresspool.c:
10597 tests: add addresspool unit test
10599 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10601 * gst/rtsp-server/Makefile.am:
10602 * gst/rtsp-server/rtsp-address-pool.c:
10603 * gst/rtsp-server/rtsp-address-pool.h:
10604 address-pool: add object to manage multicast addresses
10605 Make an object that can manage a rage of multicast addresses and ports.
10607 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10609 * gst/rtsp-server/rtsp-server.c:
10610 server: set default max-threads property
10612 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10614 * gst/rtsp-server/rtsp-media.c:
10615 media: wait for concurrent _prepare
10616 If a prepare is busy, wait for the result.
10618 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10620 * gst/rtsp-server/rtsp-media.c:
10621 media: add lock around message handler
10622 We don't want to dispatch messages while we are still processing the result of
10625 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10627 * gst/rtsp-server/rtsp-media.c:
10628 * gst/rtsp-server/rtsp-media.h:
10629 media: add lock to protect state changes
10631 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10633 * gst/rtsp-server/rtsp-stream.c:
10634 * gst/rtsp-server/rtsp-stream.h:
10635 stream: add locking
10637 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10639 * gst/rtsp-server/rtsp-stream-transport.c:
10640 * gst/rtsp-server/rtsp-stream-transport.h:
10641 * gst/rtsp-server/rtsp-stream.c:
10642 stream-transport: add keep-alive method
10644 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10646 * gst/rtsp-server/rtsp-stream-transport.c:
10647 * gst/rtsp-server/rtsp-stream-transport.h:
10648 * gst/rtsp-server/rtsp-stream.c:
10649 stream-transport: add method to handle RTP/RTCP
10650 Call new methods instead of poking into the structures directly.
10652 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10654 * gst/rtsp-server/rtsp-session-media.c:
10655 * gst/rtsp-server/rtsp-session-media.h:
10656 session-media: add locking
10658 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10660 * gst/rtsp-server/rtsp-session.c:
10661 * gst/rtsp-server/rtsp-session.h:
10662 session: add locking
10664 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10666 * gst/rtsp-server/rtsp-server.c:
10667 server: free old socket
10669 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10671 * gst/rtsp-server/rtsp-media-mapping.c:
10672 * gst/rtsp-server/rtsp-media-mapping.h:
10673 mapping: add locking
10675 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10677 * gst/rtsp-server/rtsp-media-factory.c:
10678 media-factory: add locking
10680 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10682 * gst/rtsp-server/rtsp-auth.c:
10683 * gst/rtsp-server/rtsp-auth.h:
10686 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10688 * gst/rtsp-server/rtsp-server.c:
10689 * gst/rtsp-server/rtsp-server.h:
10690 server: add max-thread property
10692 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10694 * gst/rtsp-server/rtsp-server.c:
10695 * gst/rtsp-server/rtsp-server.h:
10696 server: use a threadpool for the mainloops
10698 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10700 * gst/rtsp-server/rtsp-client.c:
10701 * gst/rtsp-server/rtsp-client.h:
10702 client: rename method
10703 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
10704 don't really create the client from the socket, we use the socket for the
10707 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10709 * gst/rtsp-server/rtsp-client.c:
10710 * gst/rtsp-server/rtsp-client.h:
10711 * gst/rtsp-server/rtsp-server.c:
10712 server: rework maincontext handling in clients
10713 Make a separate method to attach a client to a MainContext.
10714 Let the server decide in what GMainContext the client will operate and give this
10715 context to the client in attach. Then the server can later decide to use a
10716 separate thread for each client or just use the mainthread.
10718 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10720 * gst/rtsp-server/rtsp-client.c:
10721 * gst/rtsp-server/rtsp-session.c:
10722 * gst/rtsp-server/rtsp-session.h:
10723 session: move session header code in session object
10725 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10729 * examples/test-auth.c:
10730 * examples/test-launch.c:
10731 * examples/test-mp4.c:
10732 * examples/test-ogg.c:
10733 * examples/test-readme.c:
10734 * examples/test-sdp.c:
10735 * examples/test-uri.c:
10736 * examples/test-video.c:
10737 * gst/rtsp-server/rtsp-auth.c:
10738 * gst/rtsp-server/rtsp-auth.h:
10739 * gst/rtsp-server/rtsp-client.c:
10740 * gst/rtsp-server/rtsp-client.h:
10741 * gst/rtsp-server/rtsp-media-factory-uri.c:
10742 * gst/rtsp-server/rtsp-media-factory-uri.h:
10743 * gst/rtsp-server/rtsp-media-factory.c:
10744 * gst/rtsp-server/rtsp-media-factory.h:
10745 * gst/rtsp-server/rtsp-media-mapping.c:
10746 * gst/rtsp-server/rtsp-media-mapping.h:
10747 * gst/rtsp-server/rtsp-media.c:
10748 * gst/rtsp-server/rtsp-media.h:
10749 * gst/rtsp-server/rtsp-params.c:
10750 * gst/rtsp-server/rtsp-params.h:
10751 * gst/rtsp-server/rtsp-sdp.c:
10752 * gst/rtsp-server/rtsp-sdp.h:
10753 * gst/rtsp-server/rtsp-server.c:
10754 * gst/rtsp-server/rtsp-server.h:
10755 * gst/rtsp-server/rtsp-session-media.c:
10756 * gst/rtsp-server/rtsp-session-media.h:
10757 * gst/rtsp-server/rtsp-session-pool.c:
10758 * gst/rtsp-server/rtsp-session-pool.h:
10759 * gst/rtsp-server/rtsp-session.c:
10760 * gst/rtsp-server/rtsp-session.h:
10761 * gst/rtsp-server/rtsp-stream-transport.c:
10762 * gst/rtsp-server/rtsp-stream-transport.h:
10763 * gst/rtsp-server/rtsp-stream.c:
10764 * gst/rtsp-server/rtsp-stream.h:
10765 * tests/check/gst/rtspserver.c:
10766 * tests/test-cleanup.c:
10769 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10771 * gst/rtsp-server/rtsp-media.c:
10772 * gst/rtsp-server/rtsp-session-media.c:
10773 * gst/rtsp-server/rtsp-session.c:
10774 rtsp-server: added annotations to indicate type of ownership transfer of return values
10775 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10777 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
10780 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
10782 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10785 * bindings/Makefile.am:
10786 * bindings/vala/Makefile.am:
10787 * bindings/vala/gst-rtsp-server-0.10.deps:
10788 * bindings/vala/gst-rtsp-server-0.10.vapi:
10789 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10790 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10791 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10792 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10793 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10795 bindings: remove vala bindings
10796 They'll be reunited with the other GStreamer bindings
10797 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10799 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10801 * gst/rtsp-server/rtsp-client.c:
10802 * gst/rtsp-server/rtsp-session-media.c:
10803 * gst/rtsp-server/rtsp-session-media.h:
10804 * gst/rtsp-server/rtsp-stream-transport.c:
10805 * gst/rtsp-server/rtsp-stream-transport.h:
10806 rtsp: only create transport when needed
10807 Only create the StreamTransport when configured.
10809 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10811 * gst/rtsp-server/rtsp-client.c:
10812 client: small cleanup
10814 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10816 * gst/rtsp-server/rtsp-client.c:
10817 * gst/rtsp-server/rtsp-client.h:
10818 * gst/rtsp-server/rtsp-stream-transport.c:
10819 * gst/rtsp-server/rtsp-stream-transport.h:
10820 rtsp: refactor configuration of transport
10821 Move the configuration of the transport to a place where it makes
10824 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10826 * gst/rtsp-server/rtsp-client.c:
10827 client: refactor transport parsing
10829 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10831 * gst/rtsp-server/rtsp-client.c:
10832 client: refuse to change the MTU on shared media
10833 If we change the MTU of chared media, it changes for all clients.
10834 We don't want to set the MTU to something large for clients that
10837 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10839 * examples/test-mp4.c:
10840 * gst/rtsp-server/rtsp-media.c:
10841 small fixes to docs and debug
10843 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10845 * gst/rtsp-server/rtsp-stream.c:
10846 stream: transports must already have been removed
10848 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10850 * gst/rtsp-server/rtsp-media.c:
10851 * gst/rtsp-server/rtsp-stream.c:
10852 * gst/rtsp-server/rtsp-stream.h:
10853 stream: improve join and leave of the pipeline
10855 Do the cleanup properly
10858 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10860 * gst/rtsp-server/rtsp-media.c:
10861 media: move unprepare below default implementation
10862 Makes it easier to find the default implementation
10864 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10866 * gst/rtsp-server/rtsp-media.c:
10867 media: signal unprepared when we actually finish
10869 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10871 * gst/rtsp-server/rtsp-media.c:
10872 media: no need to unlock, unprepare does that when needed
10874 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10876 * docs/libs/gst-rtsp-server-sections.txt:
10877 * gst/rtsp-server/rtsp-media-factory.h:
10878 * gst/rtsp-server/rtsp-media-mapping.c:
10879 * gst/rtsp-server/rtsp-media.h:
10880 * gst/rtsp-server/rtsp-params.c:
10881 * gst/rtsp-server/rtsp-server.c:
10882 * gst/rtsp-server/rtsp-session-pool.h:
10883 * gst/rtsp-server/rtsp-session.c:
10884 * gst/rtsp-server/rtsp-session.h:
10885 * gst/rtsp-server/rtsp-stream-transport.h:
10886 * gst/rtsp-server/rtsp-stream.h:
10889 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10891 * gst/rtsp-server/rtsp-client.c:
10892 * gst/rtsp-server/rtsp-media-mapping.h:
10893 * gst/rtsp-server/rtsp-media.c:
10894 * gst/rtsp-server/rtsp-media.h:
10895 * gst/rtsp-server/rtsp-server.h:
10896 * gst/rtsp-server/rtsp-stream.c:
10897 * gst/rtsp-server/rtsp-stream.h:
10898 rtsp: fix MTU setting
10899 Fix setting of the MTU. There is no need for a vmethod.
10901 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10906 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10909 configure: bump version number after refactoring
10911 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10913 * gst/rtsp-server/Makefile.am:
10914 * gst/rtsp-server/rtsp-client.c:
10915 * gst/rtsp-server/rtsp-client.h:
10916 * gst/rtsp-server/rtsp-media-factory-uri.c:
10917 * gst/rtsp-server/rtsp-media-factory.c:
10918 * gst/rtsp-server/rtsp-media-factory.h:
10919 * gst/rtsp-server/rtsp-media.c:
10920 * gst/rtsp-server/rtsp-media.h:
10921 * gst/rtsp-server/rtsp-sdp.c:
10922 * gst/rtsp-server/rtsp-session-media.c:
10923 * gst/rtsp-server/rtsp-session-media.h:
10924 * gst/rtsp-server/rtsp-session.c:
10925 * gst/rtsp-server/rtsp-session.h:
10926 * gst/rtsp-server/rtsp-stream-transport.c:
10927 * gst/rtsp-server/rtsp-stream-transport.h:
10928 * gst/rtsp-server/rtsp-stream.c:
10929 * gst/rtsp-server/rtsp-stream.h:
10930 rtsp: massive refactoring
10931 Make GObjects from the remaining simple structures.
10932 Remove GstRTSPSessionStream, it's not needed.
10933 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
10934 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
10935 a GstRTSPStream should be transported to a client.
10936 Rename GstRTSPMediaFactory::get_element -> create_element because that
10937 more accurately describes what it does.
10938 Make nice methods instead of poking in the structures.
10939 Move some methods inside the relevant object source code.
10940 Use GPtrArray to store objects instead of plain arrays, it is more
10941 natural and allows us to more easily clean up.
10942 Move the allocation of udp ports to the Stream object. The Stream object
10943 contains the elements needed to stream the media to a client.
10944 Improve the prepare and unprepare methods. Unprepare should now undo
10945 everything prepare did. Improve also async unprepare when doing EOS on
10946 shutdown. Make sure we always unprepare correctly.
10948 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
10950 * gst/rtsp-server/rtsp-client.c:
10951 rtsp-client: Unref server address clients connected to
10952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
10954 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
10956 * gst/rtsp-server/rtsp-server.c:
10957 rtsp-server: don't ref server socket if it is NULL
10958 Fixes test_bind_already_in_use unit test again after commit 6a497440.
10959 https://bugzilla.gnome.org/show_bug.cgi?id=686644
10961 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
10963 * tests/check/Makefile.am:
10964 tests: Add libgio link dependency
10965 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
10967 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10969 * gst/rtsp-server/rtsp-media-mapping.c:
10970 * gst/rtsp-server/rtsp-media-mapping.h:
10971 rtsp-media-mapping: rename find_media vfunc to find_factory
10972 The virtual method and class method should have the same name
10973 so it is correctly represented in GIR file
10974 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10976 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10978 * gst/rtsp-server/rtsp-auth.c:
10979 * gst/rtsp-server/rtsp-client.c:
10980 * gst/rtsp-server/rtsp-media-factory-uri.c:
10981 * gst/rtsp-server/rtsp-media-factory.c:
10982 * gst/rtsp-server/rtsp-media-mapping.c:
10983 * gst/rtsp-server/rtsp-media.c:
10984 * gst/rtsp-server/rtsp-server.c:
10985 * gst/rtsp-server/rtsp-session-pool.c:
10986 * gst/rtsp-server/rtsp-session.c:
10987 rtsp-server: fixed comments and GIR annotations
10988 https://bugzilla.gnome.org/show_bug.cgi?id=680777
10990 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
10992 * gst/rtsp-server/rtsp-media-mapping.c:
10993 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
10995 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
10997 * gst/rtsp-server/rtsp-server.c:
10998 rtsp-server: allow binding on port 0 (binds on a random port)
11000 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
11002 * gst/rtsp-server/rtsp-server.c:
11003 * gst/rtsp-server/rtsp-server.h:
11004 rtsp-server: add bound-port property
11005 bound-port can be used to retrieve the port number when the server is bound on
11006 port 0, which binds on a random port.
11008 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
11010 * gst/rtsp-server/rtsp-media-factory.c:
11011 * gst/rtsp-server/rtsp-media-factory.h:
11012 rtsp-media-factory: make ::get_element overridable by GI bindings
11013 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
11014 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
11015 as the invoker for ::get_element(), making it overridable by GI generated
11018 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11020 * gst/rtsp-server/rtsp-media-factory-uri.c:
11021 rtsp-media-factory-uri: don't autoplug parsers in a loop
11022 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
11025 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11027 * gst/rtsp-server/Makefile.am:
11028 Explicitly link against gio. Fix link error on mac.
11030 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11032 * gst/rtsp-server/rtsp-session.c:
11033 session: add ttl to the transport header in SETUP
11034 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
11036 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
11038 * gst/rtsp-server/rtsp-client.c:
11039 * gst/rtsp-server/rtsp-client.h:
11040 * gst/rtsp-server/rtsp-media.c:
11041 client: Use client transport settings for multicast if allowed.
11042 This patch makes it possible for the client to send transport settings for
11043 multicast (destination && ttl). Client settings must be explicitly allowed or
11044 the server will use its own settings.
11045 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
11047 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
11050 Automatic update of common submodule
11051 From 6c0b52c to 6bb6951
11053 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
11055 * gst/rtsp-server/rtsp-client.c:
11056 rtsp-client: do not destroy the rtsp watch
11057 Don't destroy the client watch while dispatching. The rtsp watch is
11058 automatically destroyed after the rtsp watch function closed() has
11060 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
11062 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
11065 Automatic update of common submodule
11066 From 4f962f7 to 6c0b52c
11068 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
11070 * gst/rtsp-server/rtsp-media.c:
11071 media: fix check for seekability
11073 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11075 * gst/rtsp-server/rtsp-client.c:
11076 client: use more GIO
11077 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
11079 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11081 * gst/rtsp-server/rtsp-server.c:
11082 server: remove obsolete includes
11084 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11086 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
11087 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
11088 be available in "on_new_ssrc". The transports are added in
11089 gst_rtsp_media_set_state when going to PLAYING state. However,
11090 "on_new_ssrc" might be called before this happens.
11091 https://bugzilla.gnome.org/show_bug.cgi?id=683304
11093 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11095 * gst/rtsp-server/rtsp-client.c:
11096 * gst/rtsp-server/rtsp-client.h:
11097 rtsp-client: add signals for rtsp requests (fixes #683287)
11099 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
11101 * gst/rtsp-server/rtsp-client.c:
11102 * gst/rtsp-server/rtsp-client.h:
11103 add new-session signal to rtsp-client (fixes #683058)
11105 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
11108 Automatic update of common submodule
11109 From 668acee to 4f962f7
11111 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
11113 * gst/rtsp-server/rtsp-server.c:
11114 * tests/check/gst/rtspserver.c:
11115 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
11116 Do not assume that *error is set in g_socket_address_enumerator_next.
11117 Added test_bind_already_in_use unit-test.
11118 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
11120 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
11123 Automatic update of common submodule
11124 From 94ccf4c to 668acee
11126 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
11128 * gst/rtsp-server/rtsp-client.c:
11129 * gst/rtsp-server/rtsp-client.h:
11130 rtsp-client: make create_sdp virtual method
11131 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
11133 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11136 Automatic update of common submodule
11137 From 98e386f to 94ccf4c
11139 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11141 * gst/rtsp-server/rtsp-client.c:
11144 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
11146 * gst/rtsp-server/rtsp-client.c:
11147 * gst/rtsp-server/rtsp-client.h:
11148 * gst/rtsp-server/rtsp-server.c:
11149 * gst/rtsp-server/rtsp-server.h:
11150 rtsp-server: use an existing socket to establish HTTP tunnel
11151 Make it possible to transfer a socket from an HTTP server to be used as
11152 an RTSP over HTTP tunnel.
11154 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
11156 * gst/rtsp-server/rtsp-client.c:
11157 * gst/rtsp-server/rtsp-media.c:
11158 * gst/rtsp-server/rtsp-media.h:
11159 rtsp: Handle the blocksize parameter
11160 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
11162 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
11164 * tests/check/Makefile.am:
11165 * tests/check/gst/rtspserver.c:
11166 Have unit test get header from source dir, not installed dir
11167 This makes compilation of unit tests work in a build directory other
11168 than the source directory.
11169 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
11171 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
11173 * gst/rtsp-server/rtsp-media.c:
11174 rtsp-media: update for gst_element_make_from_uri() changes
11176 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
11179 * tests/Makefile.am:
11180 * tests/check/Makefile.am:
11181 * tests/check/gst/rtspserver.c:
11182 rtsp: add unit test
11183 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
11185 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
11187 * gst/rtsp-server/rtsp-media.c:
11188 rtsp-media: don't collect media stats when going to NULL
11189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
11191 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11193 * gst/rtsp-server/rtsp-client.c:
11194 client: don't leak transports
11196 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
11198 * gst/rtsp-server/rtsp-client.c:
11199 rtsp-client: free transport on no_stream in SETUP handler
11201 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
11203 * gst/rtsp-server/rtsp-client.c:
11204 rtsp-client: changed session media iteration
11205 In client_unlink_session: now don't iterate in session->medias
11206 list where items are removed by gst_rtsp_session_release_media.
11207 Instead, repeatedly remove the first item.
11209 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
11211 * gst/rtsp-server/rtsp-client.c:
11212 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
11213 GstRTSPSessionMedia is not a GObject type. When the
11214 GstRTSPSession is freed, it will free the media.
11216 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
11218 * gst/rtsp-server/rtsp-media-factory.c:
11219 factory: plug pad leak in collect_streams
11220 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
11221 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
11222 will take one reference, and the other reference will otherwise
11223 give a memory leak.
11225 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
11228 configure: suppress some warnings when debug is disabled
11229 Warnings about unused variables should be suppressed if core has the
11230 debug system disabled.
11231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11233 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11235 * docs/libs/Makefile.am:
11236 docs: fix build in uninstalled setup
11237 Include gst-plugins-base libs properly.
11239 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
11241 * docs/libs/gst-rtsp-server.types:
11242 docs: include headers defining rtsp-server object types
11243 Fixes compiler warnings during docs build.
11244 https://bugzilla.gnome.org/show_bug.cgi?id=676824
11246 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
11249 configure: Add warning flags for compiler when configuring
11250 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
11252 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11255 Automatic update of common submodule
11256 From 03a0e57 to 98e386f
11258 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11261 Automatic update of common submodule
11262 From 1fab359 to 03a0e57
11264 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
11266 * gst/rtsp-server/rtsp-client.c:
11267 client: fix GSocketAddress leak in gst_rtsp_client_accept
11268 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
11270 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11273 Automatic update of common submodule
11274 From f1b5a96 to 1fab359
11276 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11279 Automatic update of common submodule
11280 From 92b7266 to f1b5a96
11282 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11285 Automatic update of common submodule
11286 From ec1c4a8 to 92b7266
11288 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11291 Automatic update of common submodule
11292 From 3429ba6 to ec1c4a8
11294 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
11296 * gst/rtsp-server/rtsp-auth.c:
11297 * gst/rtsp-server/rtsp-client.c:
11298 * gst/rtsp-server/rtsp-media-factory-uri.c:
11299 * gst/rtsp-server/rtsp-server.c:
11300 rtsp: fix compiler warnings
11301 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
11303 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11306 Automatic update of common submodule
11307 From dc70203 to 3429ba6
11309 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11311 * gst/rtsp-server/rtsp-client.c:
11312 * gst/rtsp-server/rtsp-media-factory.c:
11313 * gst/rtsp-server/rtsp-media-factory.h:
11314 * gst/rtsp-server/rtsp-media.c:
11315 * gst/rtsp-server/rtsp-media.h:
11316 * gst/rtsp-server/rtsp-server.c:
11317 * gst/rtsp-server/rtsp-server.h:
11318 * gst/rtsp-server/rtsp-session-pool.c:
11319 * gst/rtsp-server/rtsp-session-pool.h:
11320 rtsp-server: port to new thread API
11322 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11325 Automatic update of common submodule
11326 From 6db25be to dc70203
11328 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11330 * gst/rtsp-server/rtsp-auth.c:
11331 * gst/rtsp-server/rtsp-auth.h:
11332 * gst/rtsp-server/rtsp-client.c:
11333 rtsp-server: Fix compilation and compiler warnings
11335 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11339 * gst/rtsp-server/Makefile.am:
11340 configure: Modernize autotools setup a bit
11341 Also we now only create tar.bz2 and tar.xz tarballs.
11343 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11346 Automatic update of common submodule
11347 From 464fe15 to 6db25be
11349 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11352 Automatic update of common submodule
11353 From 7fda524 to 464fe15
11355 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11358 * docs/libs/Makefile.am:
11359 * docs/version.entities.in:
11360 * gst-rtsp.spec.in:
11361 * gst/rtsp-server/Makefile.am:
11362 * pkgconfig/Makefile.am:
11363 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11364 * pkgconfig/gstreamer-rtsp-server.pc.in:
11365 * tests/Makefile.am:
11366 rtsp-server: Update versioning
11368 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11370 Merge remote-tracking branch 'origin/0.10'
11372 gst/rtsp-server/rtsp-session-pool.c
11374 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11376 * gst/rtsp-server/rtsp-session-pool.c:
11377 rtsp-server: Don't use deprecated GLib API
11379 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11381 Replace master with 0.11
11383 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11385 Merge branch 'master' into 0.11
11387 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11389 Merge branch 'master' into 0.11
11391 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
11394 A couple minor typo fixes
11396 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11398 * gst/rtsp-server/rtsp-media.c:
11399 media: fix state of the appqueue
11401 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11403 * gst/rtsp-server/rtsp-media-factory-uri.c:
11404 factory: use videoconvert
11406 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11408 * gst/rtsp-server/rtsp-media-factory-uri.c:
11409 factory: change to new style caps
11411 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11413 * gst/rtsp-server/rtsp-client.c:
11414 * gst/rtsp-server/rtsp-client.h:
11415 * gst/rtsp-server/rtsp-media-factory-uri.c:
11416 * gst/rtsp-server/rtsp-media.c:
11417 * gst/rtsp-server/rtsp-server.c:
11418 * gst/rtsp-server/rtsp-server.h:
11419 * gst/rtsp-server/rtsp-session-pool.c:
11420 rtsp-server: port to GIO
11423 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11426 configure: fix build
11428 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11431 docs: fix for gst_rtsp_server_set_port() -> _set_service()
11432 https://bugzilla.gnome.org/show_bug.cgi?id=666548
11434 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11437 * examples/Makefile.am:
11438 First rule of gst-rtsp-server club: don't talk about gst-phonon
11440 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11443 * pkgconfig/Makefile.am:
11444 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11445 * pkgconfig/gstreamer-rtsp-server.pc.in:
11446 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
11447 For consistency with all other modules.
11449 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11451 * gst/rtsp-server/rtsp-client.c:
11452 rtsp-client: update for new map API
11454 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11457 * bindings/Makefile.am:
11458 * bindings/python/Makefile.am:
11459 * bindings/python/arg-types.py:
11460 * bindings/python/codegen/Makefile.am:
11461 * bindings/python/codegen/__init__.py:
11462 * bindings/python/codegen/argtypes.py:
11463 * bindings/python/codegen/code-coverage.py:
11464 * bindings/python/codegen/codegen.py:
11465 * bindings/python/codegen/definitions.py:
11466 * bindings/python/codegen/defsparser.py:
11467 * bindings/python/codegen/docextract.py:
11468 * bindings/python/codegen/docgen.py:
11469 * bindings/python/codegen/fileprefix.override:
11470 * bindings/python/codegen/fileprefixmodule.c:
11471 * bindings/python/codegen/h2def.py:
11472 * bindings/python/codegen/mergedefs.py:
11473 * bindings/python/codegen/mkskel.py:
11474 * bindings/python/codegen/override.py:
11475 * bindings/python/codegen/reversewrapper.py:
11476 * bindings/python/codegen/scmexpr.py:
11477 * bindings/python/rtspserver-types.defs:
11478 * bindings/python/rtspserver.defs:
11479 * bindings/python/rtspserver.override:
11480 * bindings/python/rtspservermodule.c:
11481 * bindings/python/test.py:
11483 python: remove pygst-based python bindings
11484 pygi is the future, apparently.
11486 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
11489 Automatic update of common submodule
11490 From c463bc0 to 7fda524
11492 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11495 Automatic update of common submodule
11496 From 2a59016 to c463bc0
11498 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11501 Automatic update of common submodule
11502 From 0807187 to 2a59016
11504 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11507 Automatic update of common submodule
11508 From 11f0cd5 to 0807187
11510 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11512 * examples/test-auth.c:
11513 example: update for new caps
11515 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11517 * examples/test-video.c:
11518 * gst/rtsp-server/rtsp-client.c:
11519 * gst/rtsp-server/rtsp-media-factory-uri.c:
11520 * gst/rtsp-server/rtsp-media.c:
11521 * gst/rtsp-server/rtsp-media.h:
11522 * gst/rtsp-server/rtsp-session.c:
11523 * gst/rtsp-server/rtsp-session.h:
11524 rtsp-server: port some more to 0.11
11526 Remove bufferlist stuff
11527 Update for new API.
11528 Add queue before appsink now that preroll-queue-len is gone.
11529 Update for request pad changes.
11531 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11533 Merge branch 'master' into 0.11
11535 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11537 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11538 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11539 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11541 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
11543 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11544 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
11545 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11547 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11549 Merge branch 'master' into 0.11
11551 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11553 * gst/rtsp-server/rtsp-media.c:
11554 * gst/rtsp-server/rtsp-media.h:
11555 media: add a seekable boolean
11556 Maintain the seekable state with a new variable instead of reusing the
11559 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
11561 * gst/rtsp-server/rtsp-media.c:
11562 Disallow seek in live media
11564 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11566 Merge branch 'master' into 0.11
11568 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
11570 * gst/rtsp-server/rtsp-server.c:
11571 #ifdef statements for windows socket creation were missing
11573 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
11576 Automatic update of common submodule
11577 From a39eb83 to 11f0cd5
11579 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
11582 Automatic update of common submodule
11583 From 605cd9a to a39eb83
11585 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11587 Merge branch 'master' into 0.11
11589 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11591 * gst/rtsp-server/rtsp-client.c:
11592 client: use method to access property
11594 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11596 * gst/rtsp-server/rtsp-media-factory.c:
11597 * gst/rtsp-server/rtsp-media-factory.h:
11598 media-factory: add protocols property
11599 Add a property to configure the allowed protocols in the media created from the
11602 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11604 * gst/rtsp-server/rtsp-media-factory.c:
11605 * gst/rtsp-server/rtsp-media-factory.h:
11606 media-factory: add media-configure signal
11607 Add signal to allow the application to configure the media after it was created
11610 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11612 * gst/rtsp-server/rtsp-client.c:
11613 client: use method to access property
11615 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11617 * gst/rtsp-server/rtsp-media-factory.c:
11618 * gst/rtsp-server/rtsp-media-factory.h:
11619 media-factory: add protocols property
11620 Add a property to configure the allowed protocols in the media created from the
11623 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11625 * gst/rtsp-server/rtsp-media-factory.c:
11626 * gst/rtsp-server/rtsp-media-factory.h:
11627 media-factory: add media-configure signal
11628 Add signal to allow the application to configure the media after it was created
11631 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11633 Merge branch 'master' into 0.11
11635 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11637 * gst/rtsp-server/rtsp-client.c:
11638 client: use media multicast group
11640 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11642 * gst/rtsp-server/rtsp-media-factory.h:
11643 * gst/rtsp-server/rtsp-server.h:
11644 * gst/rtsp-server/rtsp-session-pool.h:
11645 * gst/rtsp-server/rtsp-session.h:
11648 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11650 * gst/rtsp-server/rtsp-client.c:
11651 * gst/rtsp-server/rtsp-sdp.h:
11652 sdp: copy and free the server ip address
11653 Copy and free the server ip address to make memory management easier later.
11655 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11657 * gst/rtsp-server/rtsp-media-factory.c:
11658 media-factory: configure multicast in media
11660 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11662 * gst/rtsp-server/rtsp-media.c:
11663 * gst/rtsp-server/rtsp-media.h:
11664 media: add property for multicast group
11665 Add a property to configure the multicast group in the media.
11666 Based on patches from Marc Leeman and Robert Krakora.
11668 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11670 * gst/rtsp-server/rtsp-media-factory.c:
11671 * gst/rtsp-server/rtsp-media-factory.h:
11672 media-factory: add property for multicast group
11673 Add a property to configure the multicast group in the media factory.
11674 Based on patches from Marc Leeman and Robert Krakora.
11676 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11678 * gst/rtsp-server/rtsp-client.c:
11679 client: do configuration of transport in one place
11680 Move the configuration of the transport destination address to where we also
11681 configure the other bits.
11683 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11685 * gst/rtsp-server/rtsp-client.c:
11686 client: use media multicast group
11688 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11690 * gst/rtsp-server/rtsp-media-factory.h:
11691 * gst/rtsp-server/rtsp-server.h:
11692 * gst/rtsp-server/rtsp-session-pool.h:
11693 * gst/rtsp-server/rtsp-session.h:
11696 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11698 * gst/rtsp-server/rtsp-client.c:
11699 * gst/rtsp-server/rtsp-sdp.h:
11700 sdp: copy and free the server ip address
11701 Copy and free the server ip address to make memory management easier later.
11703 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11705 * gst/rtsp-server/rtsp-media-factory.c:
11706 media-factory: configure multicast in media
11708 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11710 * gst/rtsp-server/rtsp-media.c:
11711 * gst/rtsp-server/rtsp-media.h:
11712 media: add property for multicast group
11713 Add a property to configure the multicast group in the media.
11714 Based on patches from Marc Leeman and Robert Krakora.
11716 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11718 * gst/rtsp-server/rtsp-media-factory.c:
11719 * gst/rtsp-server/rtsp-media-factory.h:
11720 media-factory: add property for multicast group
11721 Add a property to configure the multicast group in the media factory.
11722 Based on patches from Marc Leeman and Robert Krakora.
11724 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11726 * gst/rtsp-server/rtsp-client.c:
11727 client: do configuration of transport in one place
11728 Move the configuration of the transport destination address to where we also
11729 configure the other bits.
11731 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11733 Merge branch 'master' into 0.11
11735 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11737 * gst/rtsp-server/rtsp-client.c:
11738 client: destroy pipeline on client disconnect with no prior TEARDOWN.
11739 The problem occurs when the client abruptly closes the connection without
11740 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
11741 server is where the pipeline gets torn down. Since this handler is not called,
11742 the pipeline remains and is up and running. Subsequent clients get their own
11743 pipelines and if the do not issue TEARDOWNs then those pipelines will also
11744 remain up and running. This is a resource leak.
11746 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11748 Merge branch 'master' into 0.11
11750 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
11752 * gst/rtsp-server/rtsp-media-factory.c:
11753 * gst/rtsp-server/rtsp-media-factory.h:
11754 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
11755 For example, it can be used to retrieve source elements like appsrc, in a more
11756 convenient way than subclassing get_element.
11758 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11760 Merge branch 'master' into 0.11
11762 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
11764 * gst/rtsp-server/rtsp-server.c:
11765 rtsp-server: hold on to reference while using object
11767 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11769 * gst/rtsp-server/rtsp-media.c:
11772 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11775 configure: use unstable api
11777 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
11779 * gst/rtsp-server/rtsp-client.c:
11780 client: fix reference counting
11782 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
11784 * gst/rtsp-server/rtsp-client.c:
11785 * gst/rtsp-server/rtsp-media.c:
11786 fix compiler warnings about unused variables
11788 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
11790 * examples/test-launch.c:
11791 * examples/test-readme.c:
11792 * examples/test-uri.c:
11793 * examples/test-video.c:
11794 examples: tell rtsp uri when ready
11796 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
11799 Automatic update of common submodule
11800 From 69b981f to 605cd9a
11802 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11804 * gst/rtsp-server/rtsp-client.c:
11805 client: update for buffer API change
11807 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11809 * gst/rtsp-server/Makefile.am:
11810 Makefile.am: 0.10 => @GST_MAJORMINOR@
11812 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11814 * gst/rtsp-server/rtsp-media-factory-uri.c:
11815 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
11817 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11819 * gst/rtsp-server/.gitignore:
11820 .gitignore: 0.10 => 0.11
11822 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
11824 * gst/rtsp-server/Makefile.am:
11825 Makefile.am: 0.10 => @GST_MAJORMINOR@
11827 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11829 Merge branch 'master' into 0.11
11831 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
11834 Automatic update of common submodule
11835 From 9e5bbd5 to 69b981f
11837 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
11840 Automatic update of common submodule
11841 From fd35073 to 9e5bbd5
11843 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
11846 Automatic update of common submodule
11847 From 46dfcea to fd35073
11849 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11851 * gst/rtsp-server/rtsp-media-factory-uri.c:
11852 * gst/rtsp-server/rtsp-media.c:
11853 media: port to new caps API
11855 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11857 Merge branch 'master' into 0.11
11859 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
11861 * bindings/vala/gst-rtsp-server-0.10.vapi:
11862 Updated Vala bindings.
11863 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11865 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
11867 * gst/rtsp-server/rtsp-server.c:
11868 * gst/rtsp-server/rtsp-server.h:
11869 Add a signal for newly connected clients.
11870 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
11872 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
11874 * bindings/python/rtspserver.override:
11875 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
11877 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11879 * gst/rtsp-server/Makefile.am:
11880 * gst/rtsp-server/rtsp-client.c:
11881 * gst/rtsp-server/rtsp-funnel.c:
11882 * gst/rtsp-server/rtsp-funnel.h:
11883 * gst/rtsp-server/rtsp-media.c:
11884 rtsp-server: port to 0.11
11886 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11891 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11893 Merge branch 'master' into 0.11
11898 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11901 Automatic update of common submodule
11902 From c3cafe1 to 46dfcea
11904 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
11906 * bindings/python/Makefile.am:
11907 * bindings/python/rtspserver.defs:
11908 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
11910 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
11912 * bindings/python/arg-types.py:
11913 python bindings: add GstRTSPUrlParam
11914 Needed to implement MediaFactory virtual proxies
11916 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
11918 * bindings/python/arg-types.py:
11919 python bindings: fix returning GstRTSPUrl types
11921 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
11923 * bindings/python/arg-types.py:
11924 python bindings: add arg type for GstRTSPUrl
11926 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
11928 * bindings/python/rtspserver.defs:
11929 python bindings: fix the definition of MediaFactory.collect_stream
11931 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
11934 Automatic update of common submodule
11935 From 1ccbe09 to c3cafe1
11937 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11940 Automatic update of common submodule
11941 From 193b717 to 1ccbe09
11943 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
11946 Automatic update of common submodule
11947 From b77e2bf to 193b717
11949 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11952 build: Include lcov.mak to allow test coverage report generation
11954 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11957 Automatic update of common submodule
11958 From d8814b6 to b77e2bf
11960 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11963 Automatic update of common submodule
11964 From 6aaa286 to d8814b6
11966 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
11969 Automatic update of common submodule
11970 From 6aec6b9 to 6aaa286
11972 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
11975 autogen: wingo signed comment
11977 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
11979 * gst/rtsp-server/rtsp-session-pool.c:
11980 session: use full charset for RTSP session ID
11981 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
11982 session ID more difficult.
11983 https://bugzilla.gnome.org/show_bug.cgi?id=643812
11985 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11987 * gst/rtsp-server/Makefile.am:
11988 rtsp-server: Don't install the funnel header
11990 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
11993 Automatic update of common submodule
11994 From 1de7f6a to 6aec6b9
11996 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11999 configure: require core/base 0.10.31
12000 Needed at least for gst_plugin_feature_rank_compare_func().
12002 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
12005 Automatic update of common submodule
12006 From f94d739 to 1de7f6a
12008 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12010 * gst/rtsp-server/rtsp-media.c:
12011 media: remove more unused code
12013 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12015 * gst/rtsp-server/rtsp-media.c:
12016 * gst/rtsp-server/rtsp-media.h:
12017 media: remove duplicate filtering
12018 Remove the duplicate filtering code now that we have a released -good version.
12019 Give a warning instead.
12021 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12023 * gst/rtsp-server/rtsp-media-factory.c:
12024 * gst/rtsp-server/rtsp-media.c:
12025 media: fix default buffer size
12027 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12029 * gst/rtsp-server/rtsp-media-factory.c:
12030 * gst/rtsp-server/rtsp-media-factory.h:
12031 media-factory: add property to configure the buffer-size
12032 Add a property to configure the kernel UDP buffer size.
12034 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12036 * gst/rtsp-server/rtsp-media.c:
12037 * gst/rtsp-server/rtsp-media.h:
12038 media: add property to configure kernel buffer sizes
12039 Add a property to configure the kernel UDP buffer size.
12041 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12044 configure: set PYGOBJECT_REQ before using it
12045 https://bugzilla.gnome.org/show_bug.cgi?id=640641
12047 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12049 * docs/Makefile.am:
12050 docs: recursive into sub-directories on 'make upload'
12052 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12054 * docs/libs/gst-rtsp-server-docs.sgml:
12055 * docs/version.entities.in:
12056 docs: mention full version these docs are for, not just major-minor
12058 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12061 back to development
12063 === release 0.10.8 ===
12065 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12070 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12072 * gst/rtsp-server/rtsp-server.c:
12073 rtsp-server: clarify docs a little
12075 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12077 * gst/rtsp-server/rtsp-media.c:
12078 media: init debug category before starting thread
12080 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12082 * gst/rtsp-server/rtsp-auth.c:
12083 auth: add realm to make it more spec compliant
12085 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12087 * gst/rtsp-server/rtsp-server.c:
12088 * gst/rtsp-server/rtsp-server.h:
12089 server: add locking
12091 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12093 * examples/test-video.c:
12094 example: improve example docs a little
12096 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12098 * gst/rtsp-server/rtsp-server.c:
12099 server: ensure the watch has a ref to the server
12101 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12103 * gst/rtsp-server/rtsp-server.c:
12104 server: simpify channel function
12106 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12108 * gst/rtsp-server/rtsp-server.c:
12109 * gst/rtsp-server/rtsp-server.h:
12110 server: simplify management of channel and source
12111 We don't need to keep around the channel and source objects. Let the mainloop
12112 and the source manage the source and channel respectively.
12114 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12120 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12122 * tests/.gitignore:
12123 * tests/Makefile.am:
12124 * tests/test-cleanup.c:
12125 tests: add tests directory and cleanup test
12127 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12129 * gst/rtsp-server/rtsp-media-factory-uri.c:
12130 * gst/rtsp-server/rtsp-media-factory.c:
12131 * gst/rtsp-server/rtsp-media-mapping.c:
12132 * gst/rtsp-server/rtsp-media.c:
12133 * gst/rtsp-server/rtsp-session-pool.c:
12134 * gst/rtsp-server/rtsp-session.c:
12135 server: improve debugging in various objects
12137 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12139 * gst/rtsp-server/rtsp-server.c:
12140 server: chain up to the parent finalize
12142 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
12144 * bindings/python/rtspserver-types.defs:
12145 * bindings/python/rtspserver.defs:
12146 * bindings/python/rtspserver.override:
12147 * bindings/python/test.py:
12148 gst-rtsp-server: update python bindings
12150 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12152 * gst/rtsp-server/rtsp-client.c:
12153 client: use the response from the clientstate
12154 Create the response object only once and store in the client state.
12155 Make all methods use the state response,
12157 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12159 * gst/rtsp-server/rtsp-server.c:
12160 server: use signal to keep track of clients
12161 Keep track of all the clients that the server creates and remove them when they
12162 fire the 'closed' signal.
12164 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12166 * gst/rtsp-server/rtsp-client.c:
12167 * gst/rtsp-server/rtsp-client.h:
12168 client: emit signal when closing
12170 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12172 * examples/.gitignore:
12173 * examples/Makefile.am:
12174 * examples/test-auth.c:
12175 * examples/test-video.c:
12176 * gst/rtsp-server/rtsp-auth.c:
12177 * gst/rtsp-server/rtsp-auth.h:
12178 * gst/rtsp-server/rtsp-client.c:
12179 * gst/rtsp-server/rtsp-media-factory.c:
12180 * gst/rtsp-server/rtsp-media.c:
12181 * gst/rtsp-server/rtsp-media.h:
12182 * gst/rtsp-server/rtsp-session-pool.h:
12183 * gst/rtsp-server/rtsp-session.h:
12184 media: enable per factory authorisations
12185 Allow for adding a GstRTSPAuth on the factory and media level and check
12186 permissions when accessing the factory.
12187 Add hints to the auth methods for future more fine grained authorisation.
12188 Add example application for per factory authentication.
12190 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12192 * gst/rtsp-server/rtsp-auth.c:
12193 * gst/rtsp-server/rtsp-auth.h:
12194 * gst/rtsp-server/rtsp-client.c:
12195 * gst/rtsp-server/rtsp-client.h:
12196 * gst/rtsp-server/rtsp-params.c:
12197 * gst/rtsp-server/rtsp-params.h:
12198 rtsp-server: Pass ClientState structure arround
12199 Pass the collected information for the ongoing request in a GstRTSPClientState
12200 structure that we can then pass around to simplify the method arguments. This
12201 will also be handy when we implement logging functionality.
12203 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12205 * gst/rtsp-server/rtsp-media-factory.c:
12206 * gst/rtsp-server/rtsp-media-factory.h:
12207 media-factory: add methods to configure authorisation
12209 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12211 * gst/rtsp-server/rtsp-client.c:
12212 client: unref auth in finalize
12214 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12216 * gst/rtsp-server/rtsp-server.c:
12217 server: unref auth in finalize
12219 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12221 * docs/libs/gst-rtsp-server-docs.sgml:
12222 * docs/libs/gst-rtsp-server-sections.txt:
12223 * docs/libs/gst-rtsp-server.types:
12224 docs: add more docs
12226 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12228 * gst/rtsp-server/rtsp-server.c:
12229 * gst/rtsp-server/rtsp-server.h:
12230 server: separate create and accept
12231 Create separate create and accept methods so that subclasses can create custom
12233 Configure the server in the client object and prepare for keeping track of
12236 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12238 * gst/rtsp-server/rtsp-client.c:
12239 * gst/rtsp-server/rtsp-client.h:
12240 client: add support for setting the server.
12241 Add support for keeping a ref to the server that started this client
12244 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12246 * gst/rtsp-server/rtsp-auth.c:
12247 auth: fix memleak and add some docs
12248 Fix a memleak of the basic auth token.
12249 Add docs for the helper function
12251 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12253 * gst/rtsp-server/rtsp-auth.c:
12254 * gst/rtsp-server/rtsp-auth.h:
12255 * gst/rtsp-server/rtsp-client.c:
12256 client: delegate setup of auth to the manager
12257 Delegate the configuration of the authentication tokens to the manager object
12260 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12262 * examples/test-video.c:
12263 * gst/rtsp-server/Makefile.am:
12264 * gst/rtsp-server/rtsp-auth.c:
12265 * gst/rtsp-server/rtsp-auth.h:
12266 * gst/rtsp-server/rtsp-client.c:
12267 * gst/rtsp-server/rtsp-client.h:
12268 * gst/rtsp-server/rtsp-server.c:
12269 * gst/rtsp-server/rtsp-server.h:
12270 auth: add authentication object
12271 Add an object that can check the authorization of requests.
12272 Implement basic authentication.
12273 Add example authentication to test-video
12275 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12277 * gst/rtsp-server/rtsp-server.c:
12278 * gst/rtsp-server/rtsp-server.h:
12279 server: move includes back
12280 the includes are needed for sockaddr_in.
12282 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12284 * gst/rtsp-server/rtsp-client.c:
12285 * gst/rtsp-server/rtsp-client.h:
12286 * gst/rtsp-server/rtsp-server.c:
12287 * gst/rtsp-server/rtsp-server.h:
12288 rtsp: move network includes where they are needed
12290 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
12292 * gst/rtsp-server/rtsp-media.h:
12293 rtsp-media.h: Minor corrections in comments.
12296 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
12299 Automatic update of common submodule
12300 From e572c87 to f94d739
12302 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12306 * docs/libs/.gitignore:
12307 * examples/.gitignore:
12308 * gst/rtsp-server/.gitignore:
12311 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12313 * docs/libs/Makefile.am:
12314 docs: We don't build ps/pdf for API reference docs
12316 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12319 Automatic update of common submodule
12320 From ccbaa85 to e572c87
12322 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12325 Automatic update of common submodule
12326 From 46445ad to ccbaa85
12328 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12330 * gst/rtsp-server/Makefile.am:
12331 * gst/rtsp-server/rtsp-funnel.c:
12332 * gst/rtsp-server/rtsp-funnel.h:
12333 * gst/rtsp-server/rtsp-media.c:
12334 funnel: rename fsfunnel to rtspfunnel
12335 Rename the funnel to avoid conflicts with the farsight one.
12337 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12339 * gst/rtsp-server/Makefile.am:
12340 * gst/rtsp-server/fs-funnel.c:
12341 * gst/rtsp-server/fs-funnel.h:
12342 * gst/rtsp-server/rtsp-media.c:
12343 rtsp-media: add and use fsfunnel
12344 Add a copy of fsfunnel to the build because input-selector removed the (broken)
12345 select-all property that we need.
12347 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12349 * gst/rtsp-server/Makefile.am:
12350 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
12351 Use PKG_CONFIG_PATH specified at configure time (if any) as well
12352 for the g-ir-compiler, rather than just assuming the env var has
12355 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12362 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
12364 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12367 * gst/rtsp-server/Makefile.am:
12368 gobject-introspection: fix g-i build for uninstalled setup
12369 Requires gst-plugins-base git (> 0.10.31.2).
12371 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12373 * examples/test-uri.c:
12374 examples: add some more options and comments
12376 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12378 * gst/rtsp-server/rtsp-media-factory-uri.c:
12379 factory-uri: use right property type
12381 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12383 * gst/rtsp-server/rtsp-media-factory-uri.c:
12384 factory-uri: attempt to configure buffer-lists
12385 Attempt to configure buffer lists in the payloader for improved performance.
12387 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12389 * gst/rtsp-server/rtsp-media.c:
12390 media: attempt to configure bigger UDP buffers
12391 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
12392 send buffers with high bitrate streams.
12394 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
12396 * gst/rtsp-server/rtsp-client.c:
12397 client: use the socket length from getsockname
12398 Use the length returned by getsockname to perform the getnameinfo call because
12399 the size can depend on the socket type and platform.
12402 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12404 * docs/libs/gst-rtsp-server-docs.sgml:
12405 * docs/libs/gst-rtsp-server-sections.txt:
12406 docs: add uri factory to the docs
12408 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12410 * gst/rtsp-server/rtsp-client.c:
12411 * gst/rtsp-server/rtsp-media.h:
12414 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12416 * gst/rtsp-server/rtsp-client.c:
12417 * gst/rtsp-server/rtsp-media.c:
12418 * gst/rtsp-server/rtsp-media.h:
12419 * gst/rtsp-server/rtsp-session.c:
12420 * gst/rtsp-server/rtsp-session.h:
12421 rtsp-server: add support for buffer lists
12422 Add support for sending bufferlists received from appsink.
12425 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12427 * gst/rtsp-server/rtsp-client.c:
12428 * gst/rtsp-server/rtsp-media.c:
12429 * gst/rtsp-server/rtsp-media.h:
12430 * gst/rtsp-server/rtsp-sdp.c:
12431 media: make method to retrieve the play range
12432 Make a method to retrieve the playback range so that we can conditionally create
12433 a different range for the SDP and the PLAY requests.
12435 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12437 * gst/rtsp-server/rtsp-media.c:
12438 * gst/rtsp-server/rtsp-media.h:
12439 media: add signal to notify of state changes
12441 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12443 * gst/rtsp-server/rtsp-client.h:
12444 client: cleanup headers
12446 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12448 * gst/rtsp-server/rtsp-client.c:
12451 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12453 * gst/rtsp-server/rtsp-media-factory-uri.c:
12454 * gst/rtsp-server/rtsp-media-factory-uri.h:
12455 factory-uri: add support for gstpay
12456 Add an option to prefer gstpay over decoder + raw payloader.
12458 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12460 * gst/rtsp-server/rtsp-media-factory-uri.c:
12461 * gst/rtsp-server/rtsp-media-factory-uri.h:
12462 factory-uri: rework the autoplugger.
12463 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
12466 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12468 * gst/rtsp-server/rtsp-media-factory-uri.c:
12469 factory-uri: use better factory filter
12470 Make better payloader filter based on autoplug rank and RTP use case.
12472 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12475 Automatic update of common submodule
12476 From 169462a to 46445ad
12478 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12480 * gst/rtsp-server/rtsp-server.c:
12481 server: set SO_REUSEADDR before bind
12482 Set the SO_REUSEADDR _before_ bind() to make it actually work.
12484 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12486 * gst/rtsp-server/rtsp-media.c:
12487 * gst/rtsp-server/rtsp-media.h:
12488 media: emit prepared signal when prepared
12489 Make a 'prepared' signal and emit it when we successfully prepared the element.
12490 This signal can be used to configure the media object after it has been prepared
12493 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
12496 Automatic update of common submodule
12497 From 011bcc8 to 169462a
12499 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
12501 python an optional dependency
12502 * configure.ac: Move up valgrind and g-i checks. Make the python
12503 dependency optional, as it was before.
12505 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12507 Merge branch 'master' into 0.11
12512 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12514 * gst/rtsp-server/rtsp-media.c:
12515 media: update range when active clients changed
12516 When we changed the number of active clients, update the current range
12517 information because we want the second client connecting to a shared resource
12518 continue from where the stream currently.
12520 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12522 * gst/rtsp-server/rtsp-media-factory-uri.c:
12523 * gst/rtsp-server/rtsp-media-factory-uri.h:
12524 factory-uri: add colorspace and fix pt
12525 Rework the way we pass data to the autoplugger.
12526 When we have raw caps, plug a converter element to make pluggin to raw
12527 payloaders more successful.
12528 Make sure all dynamically plugged payloaders have a unique payload types.
12530 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12532 * examples/Makefile.am:
12533 * examples/test-uri.c:
12534 example: add example of the uri factory
12536 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12538 * gst/rtsp-server/Makefile.am:
12539 * gst/rtsp-server/rtsp-media-factory-uri.c:
12540 * gst/rtsp-server/rtsp-media-factory-uri.h:
12541 * gst/rtsp-server/rtsp-server.h:
12542 factory-uri: add a factory to stream any URI
12543 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
12546 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12548 * gst/rtsp-server/rtsp-media.c:
12549 * gst/rtsp-server/rtsp-media.h:
12550 media: ignore spurious ASYNC_DONE messages
12551 When we are dynamically adding pads, the addition of the udpsrc elements will
12552 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
12553 the real ASYNC_DONE when everything is prerolled.
12555 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12557 * gst/rtsp-server/rtsp-media-factory.c:
12558 * gst/rtsp-server/rtsp-media-factory.h:
12559 media-factory: make lock macro
12561 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
12563 * gst/rtsp-server/rtsp-client.c:
12564 rtsp-server: Remove unused variable and dead assignment
12566 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
12568 * examples/test-launch.c:
12569 * examples/test-mp4.c:
12570 * examples/test-ogg.c:
12571 * examples/test-readme.c:
12572 * examples/test-sdp.c:
12573 * examples/test-video.c:
12574 examples: Run gst-indent
12576 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
12578 * gst/rtsp-server/rtsp-client.c:
12579 * gst/rtsp-server/rtsp-media-factory.c:
12580 * gst/rtsp-server/rtsp-media-mapping.c:
12581 * gst/rtsp-server/rtsp-media.c:
12582 * gst/rtsp-server/rtsp-params.c:
12583 * gst/rtsp-server/rtsp-sdp.c:
12584 * gst/rtsp-server/rtsp-server.c:
12585 * gst/rtsp-server/rtsp-session-pool.c:
12586 * gst/rtsp-server/rtsp-session.c:
12587 rtsp-server: Run gst-indent
12588 Since it wasn't using the upstream common previously, there was no
12589 indentation check before commiting.
12591 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
12593 * gst/rtsp-server/rtsp-media-mapping.h:
12594 * gst/rtsp-server/rtsp-media.c:
12595 * gst/rtsp-server/rtsp-media.h:
12596 * gst/rtsp-server/rtsp-sdp.c:
12597 * gst/rtsp-server/rtsp-session-pool.h:
12598 * gst/rtsp-server/rtsp-session.c:
12599 * gst/rtsp-server/rtsp-session.h:
12600 rtsp-server: Some more doc fixups
12602 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12605 Makefile: Add cruft-cleaning support
12607 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12611 * docs/Makefile.am:
12612 * docs/libs/Makefile.am:
12613 * docs/libs/gst-rtsp-server-docs.sgml:
12614 * docs/libs/gst-rtsp-server-sections.txt:
12615 * docs/libs/gst-rtsp-server.types:
12616 * docs/version.entities.in:
12617 docs: Add gtk-doc build system
12619 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12621 * gst/rtsp-server/Makefile.am:
12622 Makefile.am: Use standard GIR make behaviour
12624 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
12628 autogen/configure: Bring more in sync to standard gst module behaviour
12630 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12632 * gst/rtsp-server/rtsp-media.c:
12633 media: warn and fail when gstrtpbin is not found
12635 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12638 configure: open 0.11 branch
12640 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
12644 Add common submodule
12646 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
12648 * common/ChangeLog:
12649 * common/Makefile.am:
12650 * common/c-to-xml.py:
12651 * common/check.mak:
12652 * common/coverage/coverage-report-entry.pl:
12653 * common/coverage/coverage-report.pl:
12654 * common/coverage/coverage-report.xsl:
12655 * common/coverage/lcov.mak:
12656 * common/gettext.patch:
12657 * common/glib-gen.mak:
12658 * common/gst-autogen.sh:
12659 * common/gst-xmlinspect.py:
12661 * common/gstdoc-scangobj:
12662 * common/gtk-doc-plugins.mak:
12663 * common/gtk-doc.mak:
12664 * common/m4/.gitignore:
12665 * common/m4/Makefile.am:
12666 * common/m4/README:
12667 * common/m4/as-ac-expand.m4:
12668 * common/m4/as-auto-alt.m4:
12669 * common/m4/as-compiler-flag.m4:
12670 * common/m4/as-compiler.m4:
12671 * common/m4/as-docbook.m4:
12672 * common/m4/as-libtool-tags.m4:
12673 * common/m4/as-libtool.m4:
12674 * common/m4/as-python.m4:
12675 * common/m4/as-scrub-include.m4:
12676 * common/m4/as-version.m4:
12677 * common/m4/ax_create_stdint_h.m4:
12678 * common/m4/check.m4:
12679 * common/m4/glib-gettext.m4:
12680 * common/m4/gst-arch.m4:
12681 * common/m4/gst-args.m4:
12682 * common/m4/gst-check.m4:
12683 * common/m4/gst-debuginfo.m4:
12684 * common/m4/gst-default.m4:
12685 * common/m4/gst-doc.m4:
12686 * common/m4/gst-error.m4:
12687 * common/m4/gst-feature.m4:
12688 * common/m4/gst-function.m4:
12689 * common/m4/gst-gettext.m4:
12690 * common/m4/gst-glib2.m4:
12691 * common/m4/gst-libxml2.m4:
12692 * common/m4/gst-plugindir.m4:
12693 * common/m4/gst-valgrind.m4:
12694 * common/m4/gtk-doc.m4:
12695 * common/m4/introspection.m4:
12696 * common/m4/pkg.m4:
12697 * common/mangle-tmpl.py:
12698 * common/plugins.xsl:
12700 * common/release.mak:
12701 * common/scangobj-merge.py:
12702 * common/upload.mak:
12703 common: Remove static version
12705 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
12707 * common/m4/introspection.m4:
12708 Update introspection.m4 to match usage
12710 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12714 Remove old stuff from the README
12716 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12719 back to development
12721 === release 0.10.7 ===
12723 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12728 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12730 * examples/test-ogg.c:
12731 test-ogg: remove parsers
12732 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
12733 buffers with timestamps. Using the parsers also seems to break things.
12735 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12737 * bindings/vala/gst-rtsp-server-0.10.vapi:
12738 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12739 Updated Vala bindings
12741 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
12743 * common/m4/introspection.m4:
12745 * gst/rtsp-server/Makefile.am:
12746 Added initial gobject-introspection support
12748 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12750 * gst/rtsp-server/rtsp-media-factory.c:
12751 media-factory: don't use host for shared hash key
12752 When we generate the key to share made between connections, don't include the
12753 host used to connect so that we can share media even if between clients that
12754 connected with localhost and ones with the ip address.
12756 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12758 * bindings/vala/Makefile.am:
12759 build: fix distcheck
12761 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12763 * bindings/vala/gst-rtsp-server-0.10.vapi:
12764 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12765 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12766 Update Vala bindings
12768 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
12770 * bindings/vala/Makefile.am:
12772 Fix configure checks and installation location for Vala bindings
12775 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12778 back to development
12780 === release 0.10.6 ===
12782 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12785 configure: release 0.10.6
12787 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12789 * gst/rtsp-server/rtsp-media.c:
12790 media: help the compiler a little
12792 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12794 * gst/rtsp-server/rtsp-media.c:
12795 * gst/rtsp-server/rtsp-media.h:
12796 * gst/rtsp-server/rtsp-session.c:
12797 media: cleanup media transport before freeing
12798 Cleanup the media transport data before freeing. In particular, remove the qdata
12799 from the rtpsource object.
12801 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12803 * gst/rtsp-server/rtsp-media-factory.c:
12804 * gst/rtsp-server/rtsp-media-factory.h:
12805 * gst/rtsp-server/rtsp-media.c:
12806 * gst/rtsp-server/rtsp-media.h:
12807 media-factory: add eos-shutdown property
12808 Add an eos-shutdown property that will send an EOS to the pipeline before
12809 shutting it down. This allows for nice cleanup in case of a muxer.
12812 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12814 * gst/rtsp-server/rtsp-media.c:
12815 * gst/rtsp-server/rtsp-media.h:
12816 media: use multiudpsink send-duplicates when we can
12817 If we have a new enough multiudpsink with the send-duplicates property, use this
12818 instead of doing our own filtering. Our custom filtering code should eventually
12819 be removed when we can depend on a released -good.
12821 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12823 * gst/rtsp-server/rtsp-media.c:
12824 media: don't leak destinations
12825 Refactor and cleanup the destinations array when the stream is destroyed.
12827 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12829 * gst/rtsp-server/rtsp-media.c:
12830 * gst/rtsp-server/rtsp-media.h:
12831 media: don't add udp addresses multiple times
12832 Keep track of the udp addresses we added to udpsink and never add the same udp
12833 destination twice. This avoids duplicate packets when using multicast.
12835 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12837 * gst/rtsp-server/rtsp-server.c:
12838 server: disable use of SO_LINGER
12839 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
12840 server close()s the connection.
12842 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12844 * gst/rtsp-server/rtsp-server.c:
12845 server: use 5 second linger period in SO_LINGER
12846 Wait 5 seconds before clearing the send buffers and reseting the connection with
12847 the client when we do a close. This should be enough time to get the message to
12851 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
12853 * gst/rtsp-server/rtsp-server.c:
12854 server: use SO_LINGER
12855 SO_LINGER on the socket will make sure that any pending data on the socket is
12856 flushed ASAP and that the socket connection is reset. This makes sure that the
12857 socket can be reused immediately.
12860 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12863 README: add blurb about shared media factories
12865 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
12867 * gst/rtsp-server/rtsp-media.c:
12868 Add stdlib.h for atoi()
12870 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12872 * bindings/python/Makefile.am:
12873 * bindings/vala/Makefile.am:
12874 build: distcheck fixes
12875 Fix 'make distcheck', somewhat (it still fails because it tries to
12876 install files into /usr/share/vala/vapi/ irrespective of the
12877 configured prefix).
12879 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12882 configure: bump core/base requirements to released version
12883 Makes things less confusing for people.
12885 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12888 configure: fail if GStreamer core/base requirements are not met
12890 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12892 * gst/rtsp-server/rtsp-client.c:
12893 client: improve client cleanups
12894 Make sure the session does not timeout when using TCP. We need to do this
12895 because quicktime player does not send RTCP for some reason in tunneled
12897 Refactor some cleanup code.
12900 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12902 * gst/rtsp-server/rtsp-session.c:
12903 * gst/rtsp-server/rtsp-session.h:
12904 session: add support for prevent session timeouts
12905 Add an atomix counter to prevent session timeouts when we are, for example,
12906 streaming over TCP.
12908 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12910 * gst/rtsp-server/rtsp-client.c:
12911 client: fix unlink on session timeouts
12912 When our session times out, make sure we unlink all streams in this
12914 Remove the tunnelid when closing the connection.
12916 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12918 * gst/rtsp-server/rtsp-session.c:
12919 session: small cleanups
12921 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
12923 * gst/rtsp-server/rtsp-client.c:
12924 client: handle lost_tunnel callbacks
12925 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
12926 hashtable so that we can reuse it for when the client reopens the POST
12928 Close the connection after a TEARDOWN.
12929 Make sure or watchid is cleared when the watch is removed.
12932 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12934 * gst/rtsp-server/rtsp-client.c:
12935 * gst/rtsp-server/rtsp-media.c:
12936 * gst/rtsp-server/rtsp-sdp.c:
12937 rtsp-server: add more support for multicast
12939 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12942 * gst/rtsp-server/rtsp-media.c:
12943 * gst/rtsp-server/rtsp-media.h:
12944 media: allow configuration of allowed lower transport
12946 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12948 * gst/rtsp-server/rtsp-client.h:
12949 * gst/rtsp-server/rtsp-media.c:
12950 * gst/rtsp-server/rtsp-media.h:
12951 * gst/rtsp-server/rtsp-sdp.c:
12952 * gst/rtsp-server/rtsp-sdp.h:
12953 * gst/rtsp-server/rtsp-server.c:
12954 rtsp: keep track of server ip and ipv6
12955 Keep track of how the client connected to the server and setup the udp ports
12956 with the same protocol.
12957 Copy the server ip address in the SDP so that clients can send RTCP back to
12960 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12962 * gst/rtsp-server/rtsp-session.c:
12965 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12967 * gst/rtsp-server/rtsp-client.c:
12968 client: use right size for malloc
12970 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12972 * gst/rtsp-server/rtsp-server.c:
12973 server: comment ipv6 server listening address
12975 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12977 * gst/rtsp-server/rtsp-media.c:
12978 media: allow for ipv6 sockets
12980 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12982 * gst/rtsp-server/rtsp-server.c:
12983 * gst/rtsp-server/rtsp-server.h:
12984 server: rework server part
12985 Allow setting a bind address, make sure we can deal with ipv6.
12986 Remove the port property and change with the service property.
12988 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12990 * gst/rtsp-server/rtsp-media.h:
12991 media: update comments a little
12993 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12995 * gst/rtsp-server/rtsp-client.c:
12996 client: make content-base better
12997 Use the URI formatting functions to make a content-base. Also make sure that
12998 there is a trailing / at the end.
13000 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13002 * gst/rtsp-server/rtsp-client.c:
13003 client: guard against invalid paths
13005 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13007 * examples/test-video.c:
13008 test: catch server bind errors
13010 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
13012 * gst/rtsp-server/rtsp-media.c:
13013 rtspmedia: emit "unprepared" if _prepare fails.
13014 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
13015 media object is removed from its factory's cache.
13017 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13019 * gst/rtsp-server/rtsp-media.c:
13020 media: collect media position when seek completes
13022 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
13024 * gst/rtsp-server/rtsp-client.c:
13025 client: call unlink_streams in client finalize
13028 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13030 * gst/rtsp-server/rtsp-media.c:
13031 media: limit the time to wait to something huge
13032 Avoid waiting forever but limit the timeout to 20 seconds.
13034 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13036 * gst/rtsp-server/rtsp-sdp.c:
13037 sdp: reindent and check for prepared status
13039 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13041 * gst/rtsp-server/rtsp-media.c:
13042 * gst/rtsp-server/rtsp-media.h:
13043 * gst/rtsp-server/rtsp-session.c:
13044 media: avoid doing _get_state() for state changes
13045 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
13046 until the media is prerolled or in error. This avoids doing a blocking call of
13047 gst_element_get_state() that can cause lockups when there is an error.
13050 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13052 * gst/rtsp-server/rtsp-media.c:
13055 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13057 * gst/rtsp-server/rtsp-media-factory.c:
13058 media-factory: better error handling
13059 Improve the error handling a bit.
13061 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13063 * gst/rtsp-server/rtsp-client.c:
13064 client: rework transport parsing
13065 Rework the transport parsing code so that we can ignore transports we don't
13066 support instead of just picking the first one we can parse.
13067 Configure a (for now hardcoded) destination for multicast transports.
13069 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13071 * gst/rtsp-server/rtsp-media.c:
13072 media: set multicast sink parameters
13073 Disable loop and automatic multicast join on the udpsink elements.
13074 Add some more debug info.
13075 Reset some state variables in the right place.
13076 Use the right port numbers for multicast.
13078 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13080 * gst/rtsp-server/rtsp-session.c:
13081 session: handle transport setup correctly
13082 Handle UDP, MCAST and TCP transport negotiation more correctly.
13083 Store the server session SSRC in the transport.
13085 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13087 * gst/rtsp-server/rtsp-client.c:
13088 rtsp-client: implement error_full
13089 Implement error_full to avoid some segfaults when the rtspconnection calls it.
13092 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13095 * gst/rtsp-server/rtsp-client.c:
13096 * gst/rtsp-server/rtsp-server.c:
13097 docs: update docs and comments
13099 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
13101 * gst/rtsp-server/rtsp-sdp.c:
13102 sdp: make server work better when behind a proxy
13104 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13106 * gst/rtsp-server/rtsp-client.c:
13107 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
13109 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13111 * gst/rtsp-server/rtsp-client.c:
13112 * gst/rtsp-server/rtsp-media-factory.c:
13113 * gst/rtsp-server/rtsp-media-mapping.c:
13114 * gst/rtsp-server/rtsp-media.c:
13115 * gst/rtsp-server/rtsp-server.c:
13116 * gst/rtsp-server/rtsp-session-pool.c:
13117 * gst/rtsp-server/rtsp-session.c:
13118 Use GStreamer's debugging subsystem
13120 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13122 * gst/rtsp-server/rtsp-media-factory.c:
13123 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
13125 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13128 back to development
13130 === release 0.10.5 ===
13132 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13137 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13140 configure: bump required versions
13142 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
13144 * gst/rtsp-server/rtsp-client.c:
13145 client: call weak-unref on client->sessions from finalize
13148 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13150 * gst/rtsp-server/rtsp-media.c:
13151 media: Fixed crasher where caps got unref'ed too often
13153 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13156 * pkgconfig/.gitignore:
13157 * pkgconfig/Makefile.am:
13158 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
13159 Added pkg-config file to use gst-rtsp-server uninstalled
13161 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13163 * gst/rtsp-server/rtsp-media.c:
13164 media: add some docs
13166 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
13168 * gst/rtsp-server/rtsp-client.c:
13169 rtsp: Use gst_rtsp_watch_send_message().
13170 Use gst_rtsp_watch_send_message() since the old API which used
13171 gst_rtsp_watch_queue_message() has been deprecated.
13173 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13176 back to development
13178 === release 0.10.4 ===
13180 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13185 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13187 * gst/rtsp-server/rtsp-client.c:
13188 * gst/rtsp-server/rtsp-session.c:
13189 * gst/rtsp-server/rtsp-session.h:
13190 rtsp: allocate channels in TCP mode
13191 When the client does not provide us with channels in TCP mode, allocate channels
13194 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13196 * gst/rtsp-server/rtsp-client.c:
13197 client: don't crash when tunnelid is missing
13198 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
13199 don't crash but return an error response to the client.
13202 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13204 * bindings/vala/gst-rtsp-server-0.10.vapi:
13205 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13206 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13207 bindings: update vala bindings with new method
13209 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13211 * gst/rtsp-server/rtsp-session-pool.c:
13212 * gst/rtsp-server/rtsp-session-pool.h:
13213 sessionpool: add function to filter sessions
13214 Add generic function to retrieve/remove sessions.
13216 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13219 configure: bump core/base requirements to release
13221 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13223 * gst/rtsp-server/rtsp-media.c:
13224 media: fix indentation
13226 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13228 * gst/rtsp-server/rtsp-media.c:
13229 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
13231 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13233 * gst/rtsp-server/rtsp-media.c:
13234 set state and remove elements of media in for loop
13236 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
13238 * bindings/vala/gst-rtsp-server-0.10.vapi:
13239 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13240 Added gst_rtsp_media_remove_elements function to Vala bindings
13242 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
13244 * gst/rtsp-server/rtsp-media.c:
13245 * gst/rtsp-server/rtsp-media.h:
13246 Added gst_rtsp_media_remove_elements function
13248 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
13250 * gst/rtsp-server/rtsp-media.c:
13251 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
13253 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13255 * bindings/vala/gst-rtsp-server-0.10.vapi:
13256 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13257 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13258 Updated Vala bindings
13260 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13262 * gst/rtsp-server/rtsp-media.c:
13263 * gst/rtsp-server/rtsp-media.h:
13264 Added vmethod unprepare to GstRTSPMedia
13265 The default implementation sets the state of the pipeline to GST_STATE_NULL
13267 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13269 * gst/rtsp-server/rtsp-media-factory.c:
13270 * gst/rtsp-server/rtsp-media-factory.h:
13271 Made collect_streams function public
13273 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13275 * gst/rtsp-server/rtsp-media-factory.c:
13276 * gst/rtsp-server/rtsp-media-factory.h:
13277 * gst/rtsp-server/rtsp-media.c:
13278 Added vmethod create_pipeline to GstRTSPMediaFactory
13279 The pipeline is created in this method and the GstRTSPMedia's element is added to it
13281 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13283 * gst/rtsp-server/rtsp-client.c:
13284 client: use g_source_destroy()
13285 We need to use g_source_destroy() because we might have added the source to a
13286 different main context than the default one.
13288 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13290 * gst/rtsp-server/Makefile.am:
13291 * gst/rtsp-server/rtsp-client.c:
13292 * gst/rtsp-server/rtsp-params.c:
13293 * gst/rtsp-server/rtsp-params.h:
13294 rtsp: prepare for handling GET/SET_PARAMETER
13295 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
13297 Fix return codes of handlers.
13299 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13301 * gst/rtsp-server/rtsp-media.c:
13302 media: don't leak session pads
13304 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13306 * gst/rtsp-server/rtsp-media.c:
13307 media: clean up the messages a bit
13309 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13311 * gst/rtsp-server/rtsp-sdp.c:
13312 sdp: warn and skip streams without media
13314 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13316 * bindings/vala/gst-rtsp-server-0.10.vapi:
13317 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13318 vala: Fixed typo in header file of RTSPMediaStream
13320 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13322 * gst/rtsp-server/rtsp-media.c:
13324 Fix a debug message
13325 Make dumping RTCP stats configurable
13327 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13329 * gst/rtsp-server/rtsp-media.c:
13330 media: be less verbose and leak less
13332 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13334 * gst/rtsp-server/rtsp-media.c:
13335 media: don't leak the destination address
13337 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13339 * gst/rtsp-server/rtsp-client.c:
13340 * gst/rtsp-server/rtsp-media.c:
13341 * gst/rtsp-server/rtsp-media.h:
13342 * gst/rtsp-server/rtsp-session.c:
13343 * gst/rtsp-server/rtsp-session.h:
13344 rtsp: use RTCP to keep the session alive
13345 Use the RTCP rtcp-from stats field to find the associated session and use this
13346 to keep the session alive.
13348 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13350 * gst/rtsp-server/rtsp-session.c:
13351 session: add 5sec to the real session timeout
13352 Allow the session to live 5sec longer before really timing out. This should give
13353 clients some extra time to keep the session active.
13355 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13357 * gst/rtsp-server/rtsp-client.c:
13358 client: replay OK to GET/SET_PARAMETER
13359 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
13360 so that we return OK for those requests.
13362 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13364 * gst/rtsp-server/rtsp-media.c:
13365 * gst/rtsp-server/rtsp-media.h:
13366 media: keep track of active transports
13367 Keep track of which transport is active to avoid closing the connection too
13369 Remove the destination transport also when going to NULL.
13370 Print some stats about the SDES and other RTCP messages we receive from the
13373 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13375 * examples/.gitignore:
13376 * examples/Makefile.am:
13377 * examples/test-sdp.c:
13378 example: add SDP relay example
13380 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13382 * gst/rtsp-server/rtsp-media.c:
13383 media: also count active TCP connections
13385 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13387 * gst/rtsp-server/rtsp-media-factory.c:
13388 * gst/rtsp-server/rtsp-media.c:
13389 * gst/rtsp-server/rtsp-media.h:
13390 rtsp: add support for dynamic elements
13391 Add support for dynamic elements.
13392 Don't set live pipelines back to paused.
13394 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13396 * gst/rtsp-server/rtsp-sdp.c:
13397 sdp: don't add encoding name when absent in caps
13399 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13401 * gst/rtsp-server/rtsp-client.c:
13402 client: warn when we can't do RTP-Info
13404 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13406 * gst/rtsp-server/rtsp-media-factory.c:
13407 factory: factor out the stream construction
13409 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13411 * gst/rtsp-server/rtsp-client.c:
13412 client: only add RTP-Info when we have the info
13413 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
13416 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13419 back to development
13421 === release 0.10.3 ===
13423 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13427 - Fixes a bug where it put the wrong verion in pkgconfig
13428 - Link RTP and RTCP sources
13430 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13432 * gst/rtsp-server/rtsp-media.c:
13433 * gst/rtsp-server/rtsp-media.h:
13434 media: link the RTP udpsrc to the session manager
13435 Link the RTP udpsrc and the appsrc to the session manager so that they don't
13436 shut down when the client sends a packet to open firewalls.
13438 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13440 * pkgconfig/gst-rtsp-server.pc.in:
13441 Don't use hard-coded version number in pkg-config file
13443 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13446 back to development
13448 === release 0.10.2 ===
13450 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13455 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13458 * common/m4/.gitignore:
13459 * examples/.gitignore:
13460 * pkgconfig/.gitignore:
13461 add some .gitignore files
13463 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13465 * gst/rtsp-server/rtsp-media.c:
13466 media: seek to key frames
13468 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13470 * gst/rtsp-server/rtsp-media.c:
13471 media: emit the unprepared signal by id
13472 Emit the unprepared signal by id instead of name and set the media as
13475 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13477 * gst/rtsp-server/rtsp-media.c:
13478 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
13480 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13482 * gst/rtsp-server/rtsp-server.c:
13483 Added finalize function to GstRTPSPServer to unref session pool and media mapping
13485 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
13487 * bindings/vala/gst-rtsp-server-0.10.vapi:
13488 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13489 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13490 Updated vala bindings
13492 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13494 * gst/rtsp-server/Makefile.am:
13495 * gst/rtsp-server/rtsp-client.c:
13496 * gst/rtsp-server/rtsp-media.c:
13497 server: use appsink and appsrc with the API
13498 Use the appsink/appsrc API instead of the signals for higher
13501 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13503 * examples/test-ogg.c:
13504 tests: set the payload type correctly
13506 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13508 * gst/rtsp-server/rtsp-media-factory.c:
13509 factory: connect to the unprepare signal
13510 Connect to the unprepare signal for non-reusable media so that we can remove
13511 them from the cache.
13513 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13515 * gst/rtsp-server/rtsp-media.c:
13516 * gst/rtsp-server/rtsp-media.h:
13517 media: add signal to notify of unprepare
13519 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13521 * gst/rtsp-server/rtsp-media.c:
13522 * gst/rtsp-server/rtsp-media.h:
13523 media: more work on making the media shared
13524 Add a reusable flag to medias, indicating that they can be reused after a state
13528 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13530 * examples/test-readme.c:
13531 examples: mark the example as shared for testing
13533 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13535 * gst/rtsp-server/rtsp-media.c:
13536 * gst/rtsp-server/rtsp-media.h:
13537 client: support shared media
13538 Always perform the state actions even if the target state of the pipeline is
13539 already correct, we still want to add/remove the transports when we are dealing
13541 Keep a counter of the number of active transports for a media so that we can use
13542 this to perform a state change when needed.
13543 Perform a state change of the pipeline only when the first transport was added
13544 or when there are no active transports.
13546 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
13548 * gst/rtsp-server/rtsp-client.c:
13549 client: fix refcounting crasher
13550 Don't need to remove the weak refs in the finalize methods, they are already
13551 removed in the dispose.
13552 Don't register the callback with a DestroyNofity.
13554 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13556 * gst/rtsp-server/rtsp-client.c:
13557 Fix rtsp client refcount management in TCP mode.
13558 Don't unref a client ref we never had. Fixes an unref
13559 of an already-free client object after a client
13560 teardown request for me.
13562 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
13564 * gst/rtsp-server/rtsp-session.c:
13565 docs: fix typo in API docs
13567 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13569 * gst/rtsp-server/rtsp-media.c:
13570 More seeking fixes.
13571 Keep the udp sources in playing even if we go to paused. unlock the sources when
13573 Add some more debug info.
13574 Only seek when we need to.
13575 Keep track of the position when we go to paused.
13577 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13579 * gst/rtsp-server/rtsp-client.c:
13580 * gst/rtsp-server/rtsp-media.c:
13581 * gst/rtsp-server/rtsp-media.h:
13582 Add beginnings of seeking.
13583 Parse the Range header and perform a seek on the pipeline for the requested
13584 position. It's disabled currently until I figure out what's going wrong.
13586 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13588 * gst/rtsp-server/rtsp-client.c:
13589 allow pause requests for now.
13592 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13594 * gst/rtsp-server/rtsp-client.c:
13595 Remove weak ref on the session in teardown
13596 We need to remove our weakref from the session when we do a teardown because
13597 else we close the TCP connection prematurely.
13599 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13601 * gst/rtsp-server/rtsp-client.c:
13602 * gst/rtsp-server/rtsp-client.h:
13603 * gst/rtsp-server/rtsp-session-pool.c:
13604 Do some more session cleanup
13605 Make session timeout kill the TCP connection that currently watches the
13607 Remove the client timeout property.
13609 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13611 * gst/rtsp-server/rtsp-client.c:
13612 * gst/rtsp-server/rtsp-client.h:
13613 * gst/rtsp-server/rtsp-media.c:
13614 * gst/rtsp-server/rtsp-media.h:
13615 * gst/rtsp-server/rtsp-server.c:
13616 * gst/rtsp-server/rtsp-session.c:
13617 * gst/rtsp-server/rtsp-session.h:
13619 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
13622 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13624 * examples/Makefile.am:
13625 * examples/test-launch.c:
13626 Add example server that takes launch lines
13627 Add an example server that streams any -launch line.
13629 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13631 * examples/test-readme.c:
13632 * gst/rtsp-server/rtsp-client.c:
13633 * gst/rtsp-server/rtsp-media.c:
13634 * gst/rtsp-server/rtsp-media.h:
13635 Add support for live streams
13636 Add support for live streams and ranges
13637 Start on handling TCP data transfer.
13639 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13641 * gst/rtsp-server/rtsp-media.c:
13642 Free the pipeline before other things
13645 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13647 * gst/rtsp-server/rtsp-client.c:
13648 Only free the pending tunnel if there is one
13651 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13653 * gst/rtsp-server/rtsp-client.c:
13654 * gst/rtsp-server/rtsp-client.h:
13655 * gst/rtsp-server/rtsp-media.c:
13656 rtsp-server: Add support for tunneling
13657 Add support for tunneling over HTTP.
13658 Use new connection methods to retrieve the url.
13659 Dispatch messages based on the message type instead of blindly
13660 assuming it's always a request.
13661 Keep track of the watch id so that we can remove it later.
13662 Set the media pipeline to NULL before unreffing the pipeline.
13664 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13666 * gst/rtsp-server/rtsp-client.c:
13667 * gst/rtsp-server/rtsp-client.h:
13668 Fix for channel -> watch rename in gstreamer
13669 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
13671 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13673 * gst/rtsp-server/rtsp-client.c:
13674 * gst/rtsp-server/rtsp-client.h:
13676 Use the async RTSP channels instead of spawning a new thread for each client.
13677 If a sessionid is specified in a request, fail if we don't have the session.
13679 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13681 * gst/rtsp-server/rtsp-media.c:
13682 Add better debug info
13683 Add some better debug info.
13685 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13687 * examples/test-video.c:
13689 Add support for session timeouts in the example.
13691 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13693 * gst/rtsp-server/rtsp-session-pool.c:
13694 * gst/rtsp-server/rtsp-session-pool.h:
13695 Pass GTimeVal around for performance reasons
13696 Get the current time only once and pass it around so that sessions don't have to
13697 get the current time anymore.
13698 Add experimental support for a GSource that dispatches when the session needs to
13701 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13703 * gst/rtsp-server/rtsp-session.c:
13704 * gst/rtsp-server/rtsp-session.h:
13705 Add better support for session timeouts
13706 Add a method to request the number of milliseconds when a session will timeout.
13708 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13710 * gst/rtsp-server/rtsp-media.c:
13711 * gst/rtsp-server/rtsp-media.h:
13712 Add suport for RTP manager monitoring
13713 Add the first stage in monitoring the rtp manager.
13714 Make sure we don't update the state to something we don't want.
13716 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13718 * gst/rtsp-server/rtsp-client.c:
13719 Add support for session keepalive
13720 Get and update the session timeout for all requests. get the session as early as
13723 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13725 * gst/rtsp-server/rtsp-media-factory.h:
13726 * gst/rtsp-server/rtsp-media.c:
13727 * gst/rtsp-server/rtsp-media.h:
13728 Handle media bus messages
13729 Handle media bus messages in a custom mainloop and dispatch them to the
13730 RTSPMedia objects. Let the default implementation handle some common messages.
13732 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13734 * gst/rtsp-server/rtsp-client.c:
13735 * gst/rtsp-server/rtsp-session-pool.c:
13736 * gst/rtsp-server/rtsp-session.c:
13737 Some more session timeout handling
13738 Move the session header setting code to a central place so that we always add
13739 the timeout parameter too.
13740 Handle timeouts by running the session cleanup code.
13741 Stop media before cleaning up.
13743 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13745 * gst/rtsp-server/rtsp-client.c:
13746 * gst/rtsp-server/rtsp-client.h:
13747 Add timeout property
13748 Add a timeout property ot the client and make the other properties into GObject
13751 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13753 * gst/rtsp-server/rtsp-session-pool.c:
13754 Use getters and setters in property code
13755 Use the getters and setters for the timeout property instead of locking
13758 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13760 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
13762 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13764 * gst/rtsp-server/rtsp-session-pool.c:
13765 * gst/rtsp-server/rtsp-session-pool.h:
13766 * gst/rtsp-server/rtsp-session.c:
13767 * gst/rtsp-server/rtsp-session.h:
13768 Add more timeout stuff
13769 Add method to check if a session is expired.
13770 Add method to perform cleanup on a session pool.
13772 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13774 * gst/rtsp-server/rtsp-client.c:
13775 * gst/rtsp-server/rtsp-session-pool.c:
13776 * gst/rtsp-server/rtsp-session-pool.h:
13777 * gst/rtsp-server/rtsp-session.c:
13778 * gst/rtsp-server/rtsp-session.h:
13779 Add beginnings of session timeouts and limits
13780 Add the timeout value to the Session header for unusual timeout values.
13781 Allow us to configure a limit to the amount of active sessions in a pool. Set a
13782 limit on the amount of retry we do after a sessionid collision.
13783 Add properties to the sessionid and the timeout of a session. Keep track of
13784 creation time and last access time for sessions.
13786 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13788 * gst/rtsp-server/rtsp-client.c:
13789 * gst/rtsp-server/rtsp-media.c:
13790 * gst/rtsp-server/rtsp-media.h:
13791 * gst/rtsp-server/rtsp-sdp.c:
13792 * gst/rtsp-server/rtsp-session-pool.c:
13793 * gst/rtsp-server/rtsp-session.c:
13794 * gst/rtsp-server/rtsp-session.h:
13795 Cleanup of sessions and more
13796 Fix the refcounting of media and sessions in the client. Properly clean up the
13797 session data when the client performs a teardown.
13798 Add Server header to responses.
13799 Allow for multiple uri setups in one session.
13800 Add Range header to the PLAY response and add the range attribute to the SDP
13802 Fix the session pool remove method, it used the wrong key in the hashtable. Also
13803 give the ownership of the sessionid to the session object.
13805 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13807 * gst/rtsp-server/rtsp-server.c:
13808 * gst/rtsp-server/rtsp-server.h:
13810 Rename the 'server_port' variable to simply 'port'.
13812 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13815 * gst/rtsp-server/rtsp-client.c:
13816 * gst/rtsp-server/rtsp-media.c:
13817 * gst/rtsp-server/rtsp-media.h:
13818 * gst/rtsp-server/rtsp-session.c:
13819 * gst/rtsp-server/rtsp-session.h:
13820 Rework the way we handle transports for streams
13821 Make the media accept an array of transports for the streams that we have
13822 configured for the play/pause requests.
13823 Implement server states for a client and its media.
13824 Require 0.10.22.1 (git HEAD) of gstreamer.
13826 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13828 * gst/rtsp-server/rtsp-client.c:
13829 * gst/rtsp-server/rtsp-media-factory.c:
13830 Drop const from functions dealing with urls
13831 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
13832 have the right const in them.
13834 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13836 * gst/rtsp-server/rtsp-client.c:
13837 * gst/rtsp-server/rtsp-media.c:
13838 * gst/rtsp-server/rtsp-sdp.c:
13842 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13844 * gst/rtsp-server/rtsp-client.c:
13845 * gst/rtsp-server/rtsp-media-factory.c:
13846 * gst/rtsp-server/rtsp-media.c:
13847 * gst/rtsp-server/rtsp-media.h:
13849 Don't keep a reference to the GstRTSPMedia in the stream.
13850 Free more things when freeing the GstRTSPMedia.
13852 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13855 * gst/rtsp-server/rtsp-media-factory.c:
13856 * gst/rtsp-server/rtsp-media-factory.h:
13857 * gst/rtsp-server/rtsp-media.c:
13858 * gst/rtsp-server/rtsp-media.h:
13859 * gst/rtsp-server/rtsp-server.c:
13860 * gst/rtsp-server/rtsp-server.h:
13861 More docs and small cleanups
13862 Add some more docs and update the README
13863 Cleanup some method names.
13864 Remove an unneeded idx field in the GstRTSPMediaStream
13866 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13869 * examples/Makefile.am:
13870 * examples/test-readme.c:
13871 Add a README and more example code
13872 Add a README file that contains a small introduction on how to use the server
13873 along with the example code explained in the readme.
13875 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13877 * gst/rtsp-server/rtsp-media.c:
13878 * gst/rtsp-server/rtsp-server.c:
13879 Fix some leaks and change default port
13880 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
13881 we finished the initial preroll. If we keep them locked, setting the pipeline to
13882 NULL will not stop and clean up the sources correctly.
13883 Change the default RTSP port to 8554 aka the official alternative RTSP port.
13885 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13887 * gst/rtsp-server/rtsp-session.c:
13888 * gst/rtsp-server/rtsp-session.h:
13889 Cleanups to the session object
13890 Remove some unneeded variables in the session state of a stream such as the
13891 owner media and the server transport.
13892 Get the configuration of a media stream in a session based on the media_stream
13893 in the original object instead of our cached index.
13894 Free more data in the finalize method.
13896 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13898 * gst/rtsp-server/rtsp-client.c:
13899 * gst/rtsp-server/rtsp-client.h:
13900 Cleanups and reuse media from DESCRIBE
13901 Handle thread create errors.
13902 Rename some internal methods to better match what they actually do.
13903 Handle misconfiguration of session_pool and media_mapping gracefully.
13904 Cache the DESCRIBE media and uri in the client connection and reuse them when
13905 we receive a SETUP request in the same connection for the same uri.
13906 Cleanup the client connection object.
13908 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13910 * gst/rtsp-server/rtsp-media-factory.c:
13911 * gst/rtsp-server/rtsp-media-factory.h:
13912 * gst/rtsp-server/rtsp-media.c:
13913 * gst/rtsp-server/rtsp-media.h:
13914 Add shared properties to media and factory
13915 Add the shared property to media.
13916 Implement some simple caching in the factory depending on if the media is shared
13919 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13921 * gst/rtsp-server/rtsp-client.c:
13922 Add a little comment
13923 Add some comment about the content-base header.
13925 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13927 * examples/Makefile.am:
13928 * examples/test-mp4.c:
13929 * examples/test-ogg.c:
13930 * examples/test-video.c:
13931 * gst/rtsp-server/Makefile.am:
13932 * gst/rtsp-server/rtsp-client.c:
13933 * gst/rtsp-server/rtsp-client.h:
13934 * gst/rtsp-server/rtsp-media-factory.c:
13935 * gst/rtsp-server/rtsp-media-factory.h:
13936 * gst/rtsp-server/rtsp-media.c:
13937 * gst/rtsp-server/rtsp-media.h:
13938 * gst/rtsp-server/rtsp-sdp.c:
13939 * gst/rtsp-server/rtsp-sdp.h:
13940 * gst/rtsp-server/rtsp-server.c:
13941 * gst/rtsp-server/rtsp-server.h:
13942 * gst/rtsp-server/rtsp-session.c:
13943 * gst/rtsp-server/rtsp-session.h:
13944 Reorganize things, prepare for media sharing
13945 Added various other test server examples
13946 Move the SDP message generation to a separate helper.
13947 Refactor common code for finding the session.
13948 Add content-base for realplayer compatibility
13949 Clean up request uris before processing for better vlc compatibility.
13950 Move prerolling and pipeline construction to the RTSPMedia object.
13951 Use multiudpsink for future pipeline reuse.
13953 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13956 Back to development
13959 === release 0.10.1 ===
13961 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13964 Make 0.10.1 release
13967 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13969 * bindings/vala/Makefile.am:
13971 Add more directories and files to the dist.
13973 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13975 * bindings/python/Makefile.am:
13976 * bindings/python/rtspserver.override:
13977 Fixed compile error of python bindings
13979 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13981 * bindings/vala/gst-rtsp-server-0.10.vapi:
13982 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13983 Marked values as nullable accordingly
13985 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
13987 * bindings/vala/gst-rtsp-server-0.10.vapi:
13988 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
13989 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
13990 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
13991 Updated Vala bindings
13993 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
13995 * gst/rtsp-server/rtsp-client.c:
13996 * gst/rtsp-server/rtsp-media-mapping.c:
13997 * gst/rtsp-server/rtsp-media-mapping.h:
13998 * gst/rtsp-server/rtsp-media.h:
13999 * gst/rtsp-server/rtsp-session-pool.h:
14000 Cleanups and doc updates
14001 Add some more documentation and do some minor cleanups here and there.
14003 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14005 * gst/rtsp-server/rtsp-client.c:
14006 * gst/rtsp-server/rtsp-media-factory.c:
14007 * gst/rtsp-server/rtsp-media-factory.h:
14008 * gst/rtsp-server/rtsp-media.c:
14009 * gst/rtsp-server/rtsp-media.h:
14010 * gst/rtsp-server/rtsp-session.c:
14011 * gst/rtsp-server/rtsp-session.h:
14013 Rename GstRTSPMediaBin to GstRTSPMedia
14014 Parse the request url into a GstRTSPUri object and pass this object to the
14015 various handlers and methods that require the uri.
14017 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14021 Add some more docs and remove some old code from the example.
14023 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14025 * gst/rtsp-server/rtsp-client.c:
14026 Handle state change failures better
14027 Handle state change failures better when changing the state of the pipeline to
14030 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14032 * gst/rtsp-server/rtsp-media-factory.c:
14033 * gst/rtsp-server/rtsp-media-factory.h:
14034 Make element creation more extendible
14035 Add get_element vmethod to the default MediaFactory so that subclasses can just
14036 override that method and still use the default logic for making a MediaBin from
14039 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14042 * gst/rtsp-server/Makefile.am:
14043 * gst/rtsp-server/rtsp-client.c:
14044 * gst/rtsp-server/rtsp-client.h:
14045 * gst/rtsp-server/rtsp-media-factory.c:
14046 * gst/rtsp-server/rtsp-media-factory.h:
14047 * gst/rtsp-server/rtsp-media-mapping.c:
14048 * gst/rtsp-server/rtsp-media-mapping.h:
14049 * gst/rtsp-server/rtsp-media.c:
14050 * gst/rtsp-server/rtsp-media.h:
14051 * gst/rtsp-server/rtsp-server.c:
14052 * gst/rtsp-server/rtsp-server.h:
14053 * gst/rtsp-server/rtsp-session.c:
14054 * gst/rtsp-server/rtsp-session.h:
14055 Make the server handle arbitrary pipelines
14056 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
14057 The GstMediaBin object has a handle to a bin with elements and to a list of
14058 GstMediaStream objects that this bin produces.
14059 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
14060 with methods to register and remove those mappings.
14061 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
14062 used by the server instance.
14063 Modify the example application so that it shows how to create custom pipelines
14064 attached to a specific mount point.
14065 Various misc cleanps.
14067 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14069 * gst/rtsp-server/rtsp-server.c:
14070 * gst/rtsp-server/rtsp-server.h:
14071 Allow setting a custom media factory for a server
14073 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14075 * gst/rtsp-server/rtsp-client.c:
14076 * gst/rtsp-server/rtsp-client.h:
14077 Allow setting a custom media factory for a client.
14079 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14081 * gst/rtsp-server/Makefile.am:
14082 Add Makefile entry for the media factory
14084 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14086 * gst/rtsp-server/rtsp-media-factory.c:
14087 * gst/rtsp-server/rtsp-media-factory.h:
14088 Add media factory to map urls to media pipeline objects.
14090 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14092 * gst/rtsp-server/rtsp-media.c:
14093 * gst/rtsp-server/rtsp-media.h:
14094 Add comments. Remove unused field
14096 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14098 * gst/rtsp-server/rtsp-session-pool.c:
14099 * gst/rtsp-server/rtsp-session-pool.h:
14100 Allow custom session pools to override the session id allocation algorithms Add some comments.
14102 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14104 * gst/rtsp-server/rtsp-session.h:
14107 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14109 * gst/rtsp-server/rtsp-client.c:
14110 * gst/rtsp-server/rtsp-client.h:
14111 Move the connection code in one place Add some comments
14113 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14115 * gst/rtsp-server/rtsp-server.c:
14116 * gst/rtsp-server/rtsp-server.h:
14117 Make vmethod to create and accept new clients. Add some docs.
14119 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14121 * gst/rtsp-server/rtsp-server.c:
14122 * gst/rtsp-server/rtsp-server.h:
14123 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
14125 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14127 * gst/rtsp-server/rtsp-client.c:
14128 * gst/rtsp-server/rtsp-client.h:
14129 Name the parameters more appropriately.
14131 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14133 * gst/rtsp-server/rtsp-session-pool.c:
14134 Do some more cleanup of the session pool.
14136 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14138 * gst/rtsp-server/Makefile.am:
14139 * gst/rtsp-server/rtsp-client.c:
14140 Check if return value of gst_rtsp_session_get_media is not NULL
14142 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14144 * gst/rtsp-server/Makefile.am:
14145 Install rtsp-session and rtsp-session-pool headers
14147 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14152 * bindings/python/Makefile.am:
14153 * bindings/python/arg-types.py:
14154 * bindings/python/codegen/Makefile.am:
14155 * bindings/python/codegen/__init__.py:
14156 * bindings/python/codegen/argtypes.py:
14157 * bindings/python/codegen/code-coverage.py:
14158 * bindings/python/codegen/codegen.py:
14159 * bindings/python/codegen/definitions.py:
14160 * bindings/python/codegen/defsparser.py:
14161 * bindings/python/codegen/docextract.py:
14162 * bindings/python/codegen/docgen.py:
14163 * bindings/python/codegen/fileprefix.override:
14164 * bindings/python/codegen/fileprefixmodule.c:
14165 * bindings/python/codegen/h2def.py:
14166 * bindings/python/codegen/mergedefs.py:
14167 * bindings/python/codegen/mkskel.py:
14168 * bindings/python/codegen/override.py:
14169 * bindings/python/codegen/reversewrapper.py:
14170 * bindings/python/codegen/scmexpr.py:
14171 * bindings/python/rtspserver-types.defs:
14172 * bindings/python/rtspserver.defs:
14173 * bindings/python/rtspserver.override:
14174 * bindings/python/rtspservermodule.c:
14176 Add python bindings.
14178 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14180 * bindings/Makefile.am:
14182 Don't go into python dir when requirements for python bindings are missing
14184 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14186 * bindings/Makefile.am:
14187 * bindings/vala/Makefile.am:
14189 Install Vala bindings if vala is available
14191 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14193 * bindings/vala/gst-rtsp-server-0.10.deps:
14194 * bindings/vala/gst-rtsp-server-0.10.vapi:
14195 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
14196 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
14197 * bindings/vala/packages/gst-rtsp-server-0.10.files:
14198 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
14199 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
14200 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
14201 Regenerated Vala bindings
14203 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
14205 * bindings/vala/gst-rtsp-server.vapi:
14206 * bindings/vala/packages/gst-rtsp-server.metadata:
14207 Fixed typo in included headers for vala bindings
14209 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14213 * pkgconfig/Makefile.am:
14214 * pkgconfig/gst-rtsp-server.pc.in:
14215 Added pkgconfig file
14217 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14219 * bindings/vala/gst-rtsp-server.vapi:
14220 * bindings/vala/packages/gst-rtsp-server.excludes:
14221 * bindings/vala/packages/gst-rtsp-server.gi:
14222 * bindings/vala/packages/gst-rtsp-server.metadata:
14223 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
14225 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
14227 * bindings/vala/gst-rtsp-server.vapi:
14228 * bindings/vala/packages/gst-rtsp-server.deps:
14229 * bindings/vala/packages/gst-rtsp-server.files:
14230 * bindings/vala/packages/gst-rtsp-server.gi:
14231 * bindings/vala/packages/gst-rtsp-server.metadata:
14232 * bindings/vala/packages/gst-rtsp-server.namespace:
14233 Added Vala bindings
14235 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
14237 * gst/rtsp-server/rtsp-session.c:
14238 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
14240 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14242 * examples/Makefile.am:
14243 * gst/rtsp-server/Makefile.am:
14244 Put GStreamer version in library name
14246 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14248 * examples/Makefile.am:
14249 * gst/rtsp-server/Makefile.am:
14250 Fix some issues to pass distcheck
14252 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14254 * gst/rtsp-server/rtsp-server.c:
14255 Added port property to GstRTSPServer class.
14257 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14262 * examples/Makefile.am:
14265 * gst/rtsp-server/Makefile.am:
14266 * gst/rtsp-server/rtsp-client.c:
14267 * gst/rtsp-server/rtsp-client.h:
14268 * gst/rtsp-server/rtsp-media.c:
14269 * gst/rtsp-server/rtsp-media.h:
14270 * gst/rtsp-server/rtsp-server.c:
14271 * gst/rtsp-server/rtsp-server.h:
14272 * gst/rtsp-server/rtsp-session-pool.c:
14273 * gst/rtsp-server/rtsp-session-pool.h:
14274 * gst/rtsp-server/rtsp-session.c:
14275 * gst/rtsp-server/rtsp-session.h:
14277 Split in library and example program
14279 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14281 * src/rtsp-client.h:
14282 Removed obsolete variable
14284 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
14286 * src/rtsp-client.c:
14287 * src/rtsp-client.h:
14288 Removed pipeline variable GstRTSPClient, because it's only used in one function
14290 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
14292 * src/rtsp-media.c:
14293 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
14295 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
14297 * src/rtsp-session.c:
14298 Initialize some more vars.
14300 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
14302 * src/rtsp-session.c:
14303 Initialize variable to avoid compiler warning.
14305 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
14308 Add a reasonable generic .gitignore